Re: [asterisk-users] CDR billsec greater than duration
Yes, I think so Seconds since the presumed media start E. Alex Balashov ha scritto: What is the definition of billsec, just out of curiosity? Seconds since the 200 OK from both ends / presumed media start? On Thu, 16 Aug 2007, Jaswinder Singh wrote: I made same thread few months ago and many people said that they dont have such records in plain asterisk install ( no freepbx ) . I was also using freepbx when i had this problem . Heres mine : mysql select count(*) from cdr where billsec duration; +--+ | count(*) | +--+ | 124 | +--+ this is out of 1749216 cdr records . I am also using freepbx btw . In all such cdr's duration is always 0 but billsec varies . On 15/08/07, Edoardo Serra [EMAIL PROTECTED] wrote: Hi all, I have a strange situation on a Asterisk 1.2.17 with FreePBX 2.2.1 Doing a select in the CDR table I noticed there are some calls with billsec greater than duration, duration is always 0 in those calls. How can this happens ? Am I missing something ? Tnx in advance Regards Edoardo Serra WeBRainstorm S.r.l. ___ --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 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] CDR billsec greater than duration
I noticed that fpbx calls ResetCDR on call hangup (don't know why this choice) Could it be related to that ?? Tnx E. Jaswinder Singh ha scritto: I made same thread few months ago and many people said that they dont have such records in plain asterisk install ( no freepbx ) . I was also using freepbx when i had this problem . Heres mine : mysql select count(*) from cdr where billsec duration; +--+ | count(*) | +--+ | 124 | +--+ this is out of 1749216 cdr records . I am also using freepbx btw . In all such cdr's duration is always 0 but billsec varies . On 15/08/07, *Edoardo Serra* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi all, I have a strange situation on a Asterisk 1.2.17 with FreePBX 2.2.1 Doing a select in the CDR table I noticed there are some calls with billsec greater than duration, duration is always 0 in those calls. How can this happens ? Am I missing something ? Tnx in advance Regards Edoardo Serra WeBRainstorm S.r.l. ___ --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] CDR billsec greater than duration
Hi all, I have a strange situation on a Asterisk 1.2.17 with FreePBX 2.2.1 Doing a select in the CDR table I noticed there are some calls with billsec greater than duration, duration is always 0 in those calls. How can this happens ? Am I missing something ? Tnx in advance Regards Edoardo Serra WeBRainstorm S.r.l. ___ --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] CDR billsec greater than duration
Hi all, I have a strange situation on a Asterisk 1.2.17 with FreePBX 2.2.1 Doing a select in the CDR table I noticed there are some calls with billsec greater than duration, duration is always 0 in those calls. How can this happens ? Am I missing something ? Tnx in advance Regards Edoardo Serra WeBRainstorm S.r.l. ___ --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] Queue Status
Hi, you can use an AGI to connect to asterisk manager and retrieve the info you need about the queue. Hope it helps Arun Kumar ha scritto: Hi I've few queues configured in * box is there any what that before sending call to a particular queue can we get the status of the queue that is how many agents are available in this queue (logged in, paused, busy, unavailable). thanks arun ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ing. Edoardo Serra WeBRainstorm S.r.l. Via Pio Foà 83/C 10126 - Torino Tel: +39 011 678 100 Fax: +39 011 678 275 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call waiting tone when calling a busy station?
Yehavi Bourvine +972-8-9489444 ha scritto: This is not what I meant. I want the called party to get a sign of a waiting call and answer it if he/she wants. Ok, that's an UAC option I want the caller to know that he on a waiting call (here it is customary to play a stuttered ring tone). in short - can I signal in the 183 ringing packet that this is a second call? I don't think SIP has an implementation of that My suggestion is to use a queue in which you would put callers if the called party is busy (you can check that with ome AGI scripting) You can then record a stuttered 'ring' tone and put that as background music for the queue. Queues are the best way to handle you situation even if it's not an elegant solution for playing the stuttered ring tone My 2 cents Thanks! __Yehavi: ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ing. Edoardo Serra WeBRainstorm S.r.l. Via Pio Foà 83/C 10126 - Torino Tel: +39 011 678 100 Fax: +39 011 678 275 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call waiting tone when calling a busy station?
Hello, this is a SIP phone configuration issue. You should tell the UAC to not accept a second call while the line is engaged (look for a 'Call Waiting' option in the configuration of the UAC) The UAC will send back a 486 Busy Here error code and the calling party will get a busy signal from asterisk The calling party will then play a busy tone, or Asterisk will emulate it in case of analog zaptel devices Regards Edoardo Serra WeBRainstorm S.r.l. Yehavi Bourvine +972-8-9489444 ha scritto: Hello, When dialling a SIP phone which is already in a call the caller hears a regular ringing tone and does not know that the called party is engaged in another call. Is there a supported way inside SIP to tell the calling party to play a stuttered ringing tone? Thanks! __Yehavi: ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ing. Edoardo Serra WeBRainstorm S.r.l. Via Pio Foà 83/C 10126 - Torino Tel: +39 011 678 100 Fax: +39 011 678 275 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Delay in Dial()
Hi Steve,
put a timeout in the Dial command, if the call isn't answered it
returns after the timeout has expired
e.g.:
exten = _X.,1,Dial(SIP/${EXTEN}|15)
It waits 15 secs for the call to be answered
Look at http://www.voip-info.org/wiki-Asterisk+cmd+Dial for more
informations
Regards
Edoardo
Steve Finkelstein ha scritto:
All,
Is there any syntax I can use to put a delay in two lines being dialed?
One is a SIP endpoint, the other is my cell phone. I'd like to have the
SIP phone ring for some arbitrary number of seconds before it is sent
off to the mobile phone. Using something like a Wait() within a Dial()
would be ideal.
Any suggestions?
- sf
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Tel: +39 011 678 100
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Re: [asterisk-users] automatically close a meetme
Jerry Geis ha scritto:
I am looking for a way to automatically close a meetme conference
when either a user hangs up or through an agi call?
Look at MeetMe docs.
http://www.voip-info.org/wiki-Asterisk+cmd+MeetMe
create the MeetMe with the 'x' flag and then put inside it some marked
users ('A')
when the last marked user leaves the conference is closed
Hope it helps
Edoardo
Some method that would automatically terminate the meetme.
Is there a way to do that?
Jerry
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Re: [asterisk-users] Confference function
Hi Ed Ed Nuñez ha scritto: I would like to know if anyone here knows the answer to the following question I need to implement the following conferencing feature for my agents. 1. Agent receives call from caller 2. Agent conferences a verification service No problem since here 3. After finishing the verification, agent needs to drop third party (Verification service) and continue on the line with caller. What is your Verification service ? A VoIP UA which is called for each received call ?? In this case you should kick it from the conference Here are my suggestion: - Use MeetM Web Control (http://www.voip-info.org/wiki/view/MeetMe-Web-Control) - Use MeetMe b option and write an AGI which react to DTMF pressed by the agent (Pay attention to it: http://www.voip-info.org/wiki-Asterisk+cmd+MeetMe#MoreonoptionbAGI_BACKGROUND) - Implement some dirty hack in app_meetme.c (you can define a key which kicks every user markned with 'A' option) Hope it helps Regards Edoardo Serra My problem right now is being able to disconnect the third party and keeping the caller on the line. Would this be a function of Asterisk or the SIP / IAX phone? Any comments would be appreciated. Thank you Ed Nuñez ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ing. Edoardo Serra WeBRainstorm S.r.l. Via Pio Foà 83/C 10126 - Torino Tel: +39 011 678 100 Fax: +39 011 678 275 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Trigger for unavailable SIP peer
I'm using zabbix (http://www.zabbix.com/) as a complete monitoring solution zabbix agent has the possibility to specify custom checks that are run as often as you wish (maybe an asterisk -rx sip show peers | grep UNREACHABLE | wc -l) the output of the script is sent to zabbix server which can fire actions (email, sms, etc) in a very flexible manner My 2 cents Regards C F ha scritto: Thank you all for your response, but it appears that some of you didn't understand my question. I know I can schedule a cron to check the status (I can even use asterisk -rx sip show peers | grep UNREACHABLE if I use a cron) but that is not what I want. I want either a way that just as asterisk prints to the CLI the following: Peer '120' is now UNREACHABLE! Last qualify: 118 it should also be able to trigger whatever action from a conf file or the like. Or if there is an available solution even that involves a cron job but already has all the options, so I don't have to reinvent the wheel. On 4/18/07, C F [EMAIL PROTECTED] wrote: I use qualify in sip.conf and need to setup a trigger when asterisk sees it as unreachable, so that I can either drop a call file, or send an email, or both. How can I do that? Thank you ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ing. Edoardo Serra WeBRainstorm S.r.l. Via Pio Foà 83/C 10126 - Torino Tel: +39 011 678 100 Fax: +39 011 678 275 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CallerID masking
Hello Rob, try to set che MONITOR_FILENAME as something containing the internal extension befor emasking the CID hope it helps Edoardo Rob Schall ha scritto: Hello all, I currently have all outgoing calls set to mask the caller id so it will always appear to be coming from our main number. The problem I'm having though, is with both the call detail in mysql and with the automon (recording) feature. It shows the originating number as the number I masked it to, rather than the actual person calling. How can I go about having both the destination see our main number, but our internally logging see the correct #? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ing. Edoardo Serra WeBRainstorm S.r.l. Via Pio Foà 83/C 10126 - Torino Tel: +39 011 678 100 Fax: +39 011 678 275 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using meetme like call
Hi Enrico,
you can achieve this with the G option of Dial command
Here is a quick dialplan snippet
[from-internal-custom]
exten = 4002,1,Noop(MeetMeTest Creating MeetMe ${CALLERID(num)})
exten = 4002,n,Answer()
exten = 4002,n,Set(_MEETMEROOM=${CALLERID(num)})
exten = 4002,n,Dial(SIP/XX||G(meetme-custom^s^1))
[meetme-custom]
exten = s,1,MeetMe(${MEETMEROOM},dAxqa)
exten = s,2,MeetMe(${MEETMEROOM},qdx)
When the call is estabilished, call legs are sent to meetme-custom,s,1
(caller) and meetme-custom,s,2 (called)
I used the callerid as dynamic MeetMe room
Then have a look at 'a' option of MeetMe to solve your problem related
to hangup
Hope it helps
Regards
Enrico Pasqualotto ha scritto:
hi all, I have a little question about meetme in Asterisk.
One of my client ask me that all call can, if is necessary, become
conference for 3-4 user during conversation.
I think that are 2 way for make this:
1- all call (instead if the users are only 2) are conference
2- using n-way call
(http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO)
I decide to implement the first way because for the users is the
simplest (I think).
The problem is that when user call one extension that isn't available or
not responding the first user remain in the room for all work day. :(
There's a way to make ring two phone and enter in the conference in the
same time?
Thank Enrico.
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Tel: +39 011 678 100
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[asterisk-users] Dialplan help - MeetMe and call monitoring
Hi guys,
I need to realize a sort of automatic call monitoring dialplan.
This is exactly what I need:
- A user originate a call
- When the call is bridged (or just before) I need to invite
automatically a third party to the conversation that should hear the
audio channel but not speak (it's a monitoring application for a callcenter)
The person in charge of monitoring cannot use ChanSpy or whatever
because calls are placed at random hours during the day and its
telephone should ring when he needs to listen to a call.
I was thining at using a MeetMe in which i'd put both legs of the
monitored call and the person who should hear the conversation.
Do you have other tips about that ??
Here was my first idea of dialplan to get to it.
[outgoing]
exten = _X.,1,Noop(Call monitored in MeetMe ${CALLERID(num)})
exten = _X.,n,Answer()
exten = _X.,n,Set(_MEETMEROOM=${CALLERID(num)})
exten = _X.,n,Wait(1)
exten = _X.,n,Dial(SIP/trunk/${EXTEN}|G(call-third-party^s^1))
[invite-third-party]
exten = s,1,MeetMe(${MEETMEROOM},dAxqa)
exten = s,2,Dial(SIP/extensionformonitoring|G(bridge-all^s^1))
[bridge-all]
exten = s,1,MeetMe(${MEETMEROOM},qdx)
exten = s,2,MeetMe(${MEETMEROOM},mqdx)
This setup is not working because I cannot call a Dial again on a
bridged channel
Here is what I get on Asterisk CLI
== Starting SIP/180-108c94b0 at call-third-party,s,2 failed so
falling back to exten 's'
== Starting SIP/180-108c94b0 at call-third-party,s,2 still failed so
falling back to context 'default'
Do you have some idea to achieve this kind of result ?
Tnx in advance
Regards
Edoardo Serra
WeBRainstorm S.r.l.
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[asterisk-users] Dialplan help - MeetMe (or ChannelRedirect) and call monitoring
Hi guys,
I need to realize a sort of automatic call monitoring dialplan.
This is exactly what I need:
- A user originate a call
- When the call is bridged (or just before) I need to invite
automatically a third party to the conversation that should hear the
audio channel but not speak (it's a monitoring application for a
callcenter)
The person in charge of monitoring cannot use ChanSpy or whatever
because calls are placed at random hours during the day and its
telephone should ring when he needs to listen to a call.
I was thining at using a MeetMe in which i'd put both legs of the
monitored call and the person who should hear the conversation.
Do you have other tips about that ??
Here was my first idea of dialplan to get to it.
[outgoing]
exten = _X.,1,Noop(Call monitored in MeetMe ${CALLERID(num)})
exten = _X.,n,Answer()
exten = _X.,n,Set(_MEETMEROOM=${CALLERID(num)})
exten = _X.,n,Wait(1)
exten = _X.,n,Dial(SIP/trunk/${EXTEN}|G(call-third-party^s1))
[invite-third-party]
exten = s,1,MeetMe(${MEETMEROOM},dAxqa)
exten = s,2,Dial(SIP/extensionformonitoring|G(bridge-all^s1))
[bridge-all]
exten = s,1,MeetMe(${MEETMEROOM},qdx)
exten = s,2,MeetMe(${MEETMEROOM},mqdx)
This setup is not working because I cannot call a Dial again on a
bridged channel
Here is what I get on Asterisk CLI
== Starting SIP/180-108c94b0 at call-third-party,s,2 failed so falling
back to exten 's'
== Starting SIP/180-108c94b0 at call-third-party,s,2 still failed so
falling back to context 'default'
Do you have some idea to achieve this kind of result ?
Maybe I can use ChannelRedirect from Asterisk 1.4 ?
Cna you give me a hint on that ?
Tnx in advance
Regards
Edoardo Serra
WeBRainstorm S.r.l.
--
Ing. Edoardo Serra
WeBRainstorm S.r.l.
Via Pio Foà 83/C
10126 - Torino
Tel: +39 011 678 100
Fax: +39 011 678 275
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Re: [asterisk-users] Maximum retries exceeded on transmission
Same to me !! Calls from OpenSER to Asterisk It happens only with Asterisk versions = 1.2.14 I'm going to capture some traffic Tnx for help Regards Alex Balashov ha scritto: Joao, It sounds like the proxy is not acknowledging the Asterisk's processing of the INVITE, but I could be wrong. It would be helpful to supply a packet capture between OpenSER and Asterisk so we could see the setup flow. Thanks, -- Alex On Tue, 10 Apr 2007, Joao Pereira said something to this effect: Hello My asterisk is receiving calls from OpenSER but all calls hangup in 20 seconds. This only happens because Im using Asterisk2Billing's AGI (without A2Billing it doesnt hang up). does someone knows whats the problem?? Here is my Asterisk debug: (xxx.xxx.xxx.xxx - the phone's IP) Apr 10 02:03:02 WARNING[6996]: res_musiconhold.c:508 monmp3thread: Unable to spawn mp3player Apr 10 02:04:18 NOTICE[8349]: rtp.c:331 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: xxx.xxx.xxx.xxx Apr 10 02:04:19 WARNING[7007]: chan_sip.c:1228 retrans_pkt: Maximum retries exceeded on transmission [EMAIL PROTECTED] for seqno 12282 (Critical Response) Apr 10 02:04:19 WARNING[7007]: chan_sip.c:1245 retrans_pkt: Hanging up call [EMAIL PROTECTED] - no reply to our critical packet. Apr 10 02:06:01 NOTICE[8360]: rtp.c:331 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: xxx.xxx.xxx.xxx Thanks for the help Regards Joao Pereira ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ing. Edoardo Serra WeBRainstorm S.r.l. Via Pio Foà 83/C 10126 - Torino Tel: +39 011 678 100 Fax: +39 011 678 275 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Sponsored development - Monodirectional audio handling
Philipp Kempgen ha scritto: Edoardo Serra wrote: The purpose of thisi implementation is to deal with some carriers that give us the call as ANSWERED when the called party is still ringing. Our billing software is billing the user (and the carrier is billing us) even with unsuccessful calls. How about you try a different carrier or send your lawyer? This could be a good idea but it happens with many carriers and many destinations It's a technology problem, if you terminate on an FXO (and many carriers use FXO for certain countries) you cannot know when the call gets answered. Every time we notice this kind of situation our carriers try to solve it changing their termination. Regards Edoardo Regards, Philipp ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: SIP/IAX peers UNREACHABLE and audio loss
Hi guys, I think I got the point of the problem. I guess it's related to a lock in res_perl (which we use to do lcr, billing, ecc...) I'll open another thread for that Tnx for hep Regards Edoardo Serra WeBRainstorm S.r.l. Edoardo Serra ha scritto: Hi all, I'm having a problem with some Asterisk servers interconnected with each other using IAX (I also tried with SIP without solving the problem) Sometimes, with apparently no reason, some peers become UNREACHABLE (I have qualify=yes in iax.conf) and REACHABLE again as soon as another qualify test is made. Our users are also complaining about audio loss during their calls, apparently randomly, everything goes ok for days and bad for another few days. I strongly believe the 2 problems are strictly related because in the logs I see REACHABLE / UNREACHABLE messages only for certains days without regularity. The days in wich i see a lot of messages are exactly the days with most of complaint about audio loss I just noticed that timestamps of the logs (REACHABLE / UNREACHABLE) are quite always during business hours, this makes me think at somewhat related to load (cpu load, badwidth load, calls load, etc...) But, looking at hardware specs of our lan, servers and average load I don't think they are over-stressed. Our servers are all: 2 x Intel(R) Xeon(TM) CPU 3.20GHz 1 GB RAM 2 x IDE HDDs Software RAID 1 Asterisk 1.2.13 with res_perl Gentoo Linux Some of them has a Sangoma card connected with an E1 Most ot these are on the same LAN, interconnected with a 1 GB switch (I don't think it should be a bandwidth problem). Load averages of these server is varying from 0.5 to 1.0 (I guess it should be ok) On each server we don't have more than 50 concurrent calls (bridged SIP - IAX2 or IAX2 - ZAP) Used codec is mostly G729 Sometimes on asterisk cli i see some messages like Avoided initial deadlock for '0x9fd130', 10 retries! I don't know if it could be somehow related. Someone of you can point me in the right direction ? Tnx in advance Regards Ing. Edoardo Serra WeBRainstorm S.r.l. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.2 and res_perl - lock that leads to weird behaviour
Hi guys, as I wrote in a previous thread I was experiencing dropped audio (apparently randomly) and SIP + IAX peers getting REACHABLE / UNREACHABLE without reason, servers were in the same LAN. Investingating deeply in the problem I also noticed that 'show channels' command on the CLI, sometimes were returning strange results, for example it wasn0t showing some channels I was sure were active. Looking at our DB's log, I also notited there were a race condition which could lock a query for a long time (up to 30 secs) I don't want to annoy you wit DB issues explaining why it happened My Asterisk dialplan, when a user try to place a call, make a query to our db (through res_perl) to check some parameters, among them the user's credit to set the maximum duration of a call. If that perl script does not end in a reasonable time (I cannot tell how much is reasonable, but the 30secs due to the lock were surely too many) and other users try to call (and also their queries get locked) Asterisk begins causing weird problems. (I saw on res_perl documentation that it acquires some lock in asterisk during scripts execution but I didn't imagine that locks could affect the whole Asterisk box) Common problems in these cases are peers qualified as UNREACHABLE, dropped audio (sometimes in both directions, sometimes in just one direction), channels missing in 'show channels', etc I solved the race condition at db level and problem have magically disappeared but I'd like to go deep in the problem, I wouldn't like that o happen again because of a slow query or sloq execution of a perl script (it could happen for a lot of reasons) Someone can help with that ? Sorry for the crosspost but I think also asterisk-devel could be involved in it. Tnn in advance for help Regards Edoardo Serra WeBRainstorm S.r.l. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.2 and res_perl - lock that leads to weird behaviour
Hi guys, as I wrote in a previous thread I was experiencing dropped audio (apparently randomly) and SIP + IAX peers getting REACHABLE / UNREACHABLE without reason, servers were in the same LAN. Investingating deeply in the problem I also noticed that 'show channels' command on the CLI, sometimes were returning strange results, for example it wasn0t showing some channels I was sure were active. Looking at our DB's log, I also notited there were a race condition which could lock a query for a long time (up to 30 secs) I don't want to annoy you wit DB issues explaining why it happened My Asterisk dialplan, when a user try to place a call, make a query to our db (through res_perl) to check some parameters, among them the user's credit to set the maximum duration of a call. If that perl script does not end in a reasonable time (I cannot tell how much is reasonable, but the 30secs due to the lock were surely too many) and other users try to call (and also their queries get locked) Asterisk begins causing weird problems. (I saw on res_perl documentation that it acquires some lock in asterisk during scripts execution but I didn't imagine that locks could affect the whole Asterisk box) Common problems in these cases are peers qualified as UNREACHABLE, dropped audio (sometimes in both directions, sometimes in just one direction), channels missing in 'show channels', etc I solved the race condition at db level and problem have magically disappeared but I'd like to go deep in the problem, I wouldn't like that o happen again because of a slow query or sloq execution of a perl script (it could happen for a lot of reasons) Someone can help with that ? Sorry for the crosspost but I think also asterisk-devel could be involved in it. Tnn in advance for help Regards Edoardo Serra WeBRainstorm S.r.l. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sponsored development - Monodirectional audio handling
Hi Guys, we're needing a special implementation on Asterisk Our intention is to contribute the development and share back the code to Asterisk community Here is what we need: - An option to Asterisk Dial command which, if used, when calls is answered gives monodirectional audio (Caller should hear the called party but not vice-versa) - A DTMF sequence (maybe handled in features.conf) for the Caller to start to have bidirectional audio - When the Callers makes the audio 'bidirectional' an Event should be generated so that we can see it from the manager API The purpose of thisi implementation is to deal with some carriers that give us the call as ANSWERED when the called party is still ringing. Our billing software is billing the user (and the carrier is billing us) even with unsuccessful calls. This way we can start billing when the user press the DTMF sequence to unlock audio (even if carriers bill us wrongly) Someone wants to help ?? Regards Edoardo Serra WeBRainstorm S.r.l. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Sponsored development - Monodirectional audio handling
Salvatore Giudice ha scritto: You could put a bounty on this. You may find someone who will be willing to write this for money. My Bounty for that feature is 500 USD -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (702) 979-2906 Fax: (212) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Edoardo Serra Sent: Saturday, March 31, 2007 11:42 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Sponsored development - Monodirectional audio handling Hi Guys, we're needing a special implementation on Asterisk Our intention is to contribute the development and share back the code to Asterisk community Here is what we need: - An option to Asterisk Dial command which, if used, when calls is answered gives monodirectional audio (Caller should hear the called party but not vice-versa) - A DTMF sequence (maybe handled in features.conf) for the Caller to start to have bidirectional audio - When the Callers makes the audio 'bidirectional' an Event should be generated so that we can see it from the manager API The purpose of thisi implementation is to deal with some carriers that give us the call as ANSWERED when the called party is still ringing. Our billing software is billing the user (and the carrier is billing us) even with unsuccessful calls. This way we can start billing when the user press the DTMF sequence to unlock audio (even if carriers bill us wrongly) Someone wants to help ?? Regards Edoardo Serra WeBRainstorm S.r.l. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: SIP/IAX peers UNREACHABLE and audio loss
Hi all, I made some tests under heavy network load generated artificially moving files form server to server I noticed a 3% packet loss in ping -f response form server involved in big data transfer (1 GB files through http) I changed the network switch with a Cisco Catalyst 2950 and the packet loss with pings disapperead but the problem with REACHABLE / UNREACHABLE peers remains... I did one more simple test While Asterisk is stating the peer is UNREACHABLE I can ping (even -f) it without problem and without packet loss. Could it be a problem in Asterisk ? I'm using 1.2.13 on a gentoo Kernel 2.6.20 Tnx again for help Edoardo Edoardo Serra ha scritto: Hi all, I'm having a problem with some Asterisk servers interconnected with each other using IAX (I also tried with SIP without solving the problem) Sometimes, with apparently no reason, some peers become UNREACHABLE (I have qualify=yes in iax.conf) and REACHABLE again as soon as another qualify test is made. Our users are also complaining about audio loss during their calls, apparently randomly, everything goes ok for days and bad for another few days. I strongly believe the 2 problems are strictly related because in the logs I see REACHABLE / UNREACHABLE messages only for certains days without regularity. The days in wich i see a lot of messages are exactly the days with most of complaint about audio loss I just noticed that timestamps of the logs (REACHABLE / UNREACHABLE) are quite always during business hours, this makes me think at somewhat related to load (cpu load, badwidth load, calls load, etc...) But, looking at hardware specs of our lan, servers and average load I don't think they are over-stressed. Our servers are all: 2 x Intel(R) Xeon(TM) CPU 3.20GHz 1 GB RAM 2 x IDE HDDs Software RAID 1 Asterisk 1.2.13 with res_perl Gentoo Linux Some of them has a Sangoma card connected with an E1 Most ot these are on the same LAN, interconnected with a 1 GB switch (I don't think it should be a bandwidth problem). Load averages of these server is varying from 0.5 to 1.0 (I guess it should be ok) On each server we don't have more than 50 concurrent calls (bridged SIP - IAX2 or IAX2 - ZAP) Used codec is mostly G729 Sometimes on asterisk cli i see some messages like Avoided initial deadlock for '0x9fd130', 10 retries! I don't know if it could be somehow related. Someone of you can point me in the right direction ? Tnx in advance Regards Ing. Edoardo Serra WeBRainstorm S.r.l. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Two or More Bri Cards
I Always had very bad experiences with 2 HFC cards in the same box I strongly suggest you to use a dual port card Regards Edoardo Farooq Ahmed ha scritto: hi all we want to use Two single port Bri cards in Trixbox. Any idea which card is having good support and performance repotation especially when using two or more in Trixbox. Regards farooq ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: SIP/IAX peers UNREACHABLE and audio loss
Hi Rajeev, Rajeev Natarajan ha scritto: Well, we have add similar issues - do you use a media gateway /.IP Phones / softphones as your extensions? The problem happens mainly between server with Asterisks ! We were running Audiocodes and for some reason (I suspect a poor ethernet switch), when there are more than 15 people using the line, Audiocodes will not respond to a qualify and asterisk will drop the call. Does Asterisk drop the line if the peer becomes UNREACHABLE ? Even if RTP is still flowing ?? Turned off qualify (removed qualify=yes) and still keeping fingers crossed things seem fine. I'll give it a try Tnx for help Edoardo Rajeev On 3/23/07, *Edoardo Serra* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi all, I'm having a problem with some Asterisk servers interconnected with each other using IAX (I also tried with SIP without solving the problem) Sometimes, with apparently no reason, some peers become UNREACHABLE (I have qualify=yes in iax.conf) and REACHABLE again as soon as another qualify test is made. Our users are also complaining about audio loss during their calls, apparently randomly, everything goes ok for days and bad for another few days. I strongly believe the 2 problems are strictly related because in the logs I see REACHABLE / UNREACHABLE messages only for certains days without regularity. The days in wich i see a lot of messages are exactly the days with most of complaint about audio loss I just noticed that timestamps of the logs (REACHABLE / UNREACHABLE) are quite always during business hours, this makes me think at somewhat related to load (cpu load, badwidth load, calls load, etc...) But, looking at hardware specs of our lan, servers and average load I don't think they are over-stressed. Our servers are all: 2 x Intel(R) Xeon(TM) CPU 3.20GHz 1 GB RAM 2 x IDE HDDs Software RAID 1 Asterisk 1.2.13 with res_perl Gentoo Linux Some of them has a Sangoma card connected with an E1 Most ot these are on the same LAN, interconnected with a 1 GB switch (I don't think it should be a bandwidth problem). Load averages of these server is varying from 0.5 to 1.0 (I guess it should be ok) On each server we don't have more than 50 concurrent calls (bridged SIP - IAX2 or IAX2 - ZAP) Used codec is mostly G729 Sometimes on asterisk cli i see some messages like Avoided initial deadlock for '0x9fd130', 10 retries! I don't know if it could be somehow related. Someone of you can point me in the right direction ? Tnx in advance Regards Ing. Edoardo Serra WeBRainstorm S.r.l. ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- 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 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: RE : SIP/IAX peers UNREACHABLE and audio loss
Hi Francois, [EMAIL PROTECTED] ha scritto: Hi men, I have already encountered some issue like this with few switches (very known great brand) which doesn't like VoIP traffic ! I also have switches of a very known great brand !! It was so strange to me that I didn't consider a network problem... Check by drectly connected the VoIP equipment - if you can - with temporary long Ethernet cables bypassing the tested switch to see what happens in this case. I'd try to bypass the switch someway but every server neeeds to have its own public ip address.. I'll put an RTP proxy somewhere... You can also tell to qualify with a longer delay, but this could not help in case of regulary frames losses. What about turning qualify off ? Do you think taht Asterisk is stopping RTP when it loose a qualify packet ? Or is the RTP traffic itself that is lost by the switches ? Good luck ! It couldn't be more appropriate... Tnx for help ;) Edoardo Francois BERGERET, France. -Message d'origine- *De :* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *De la part de* Rajeev Natarajan *Envoyé :* samedi 24 mars 2007 08:14 *À :* Asterisk Users Mailing List - Non-Commercial Discussion *Objet :* Re: [asterisk-users] SIP/IAX peers UNREACHABLE and audio loss Well, we have add similar issues - do you use a media gateway /.IP Phones / softphones as your extensions? We were running Audiocodes and for some reason (I suspect a poor ethernet switch), when there are more than 15 people using the line, Audiocodes will not respond to a qualify and asterisk will drop the call. Turned off qualify (removed qualify=yes) and still keeping fingers crossed things seem fine. Rajeev On 3/23/07, *Edoardo Serra* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi all, I'm having a problem with some Asterisk servers interconnected with each other using IAX (I also tried with SIP without solving the problem) Sometimes, with apparently no reason, some peers become UNREACHABLE (I have qualify=yes in iax.conf) and REACHABLE again as soon as another qualify test is made. Our users are also complaining about audio loss during their calls, apparently randomly, everything goes ok for days and bad for another few days. I strongly believe the 2 problems are strictly related because in the logs I see REACHABLE / UNREACHABLE messages only for certains days without regularity. The days in wich i see a lot of messages are exactly the days with most of complaint about audio loss I just noticed that timestamps of the logs (REACHABLE / UNREACHABLE) are quite always during business hours, this makes me think at somewhat related to load (cpu load, badwidth load, calls load, etc...) But, looking at hardware specs of our lan, servers and average load I don't think they are over-stressed. Our servers are all: 2 x Intel(R) Xeon(TM) CPU 3.20GHz 1 GB RAM 2 x IDE HDDs Software RAID 1 Asterisk 1.2.13 with res_perl Gentoo Linux Some of them has a Sangoma card connected with an E1 Most ot these are on the same LAN, interconnected with a 1 GB switch (I don't think it should be a bandwidth problem). Load averages of these server is varying from 0.5 to 1.0 (I guess it should be ok) On each server we don't have more than 50 concurrent calls (bridged SIP - IAX2 or IAX2 - ZAP) Used codec is mostly G729 Sometimes on asterisk cli i see some messages like Avoided initial deadlock for '0x9fd130', 10 retries! I don't know if it could be somehow related. Someone of you can point me in the right direction ? Tnx in advance Regards Ing. Edoardo Serra WeBRainstorm S.r.l. ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- 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 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: RE : SIP/IAX peers UNREACHABLE and audio loss
Martin Joseph ha scritto: The fact that qualify fails means you have a network issue. The same reason qualify fails (ie servers can't communicate) is the reason your users are experiencing quality issues in call. It was also my first though, but my LAN is very SIMPLE, so I was wondering if something else could cause the problem. turn off Qualify isn't going to fix anything IMO. It's just going to hide it from you. You're probably right, but it depends on Asterisk internals (which I don't know well). If Asterisk would stop to send RTP audio when just a qualify packet get lost it can make the situation worst. If the asterisk servers are all on your LAN then the network issue should be easily fixable. It should, but my LAN is very simple... I have a 10/100 Mbit switch with no more than 15 servers on it. Traffic on the LAN is not heavy even if the time of the day I see in the logs make me think it could be an issue related to network load trafic Anyhow I'll try to generate some heavy traffic on the LAN to see if it could be related to that. I also noticed that this problem began to happen when I upgraded my Asterisk to 1.2, but it can be a concidence. Do you think it could be related to bugs in ethernet drivers, kernel or whatever at the OS level ?? If the Asterisk servers are at remote locations and are using public internet, you might have problems resolving this completely. We have some Asterisk spread all over the public Internet, but firstly we should solve this problem at a LAN level Tnx for attention Regards Marty ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: RE : Re: RE : SIP/IAX peers UNREACHABLE and audio loss
[EMAIL PROTECTED] ha scritto: Have you taken care of any eventual IRQ sharing ? I don't think so. (how cuold I detect it ? ) Servers are not self assembled but brand machines They have no other pci cards (some of them have, but the problem happens also between server with no added pci cards) this is my /proc/interrupts # cat /proc/interrupts CPU0 CPU1 CPU2 CPU3 0: 486589652 44037074 667253957 76390228IO-APIC-edge timer 8: 2 0 0 0IO-APIC-edge rtc 9: 0 0 0 0 IO-APIC-level acpi 10: 0 0 0 0 IO-APIC-level ohci_hcd:usb1 14: 22211 0 26603304 0IO-APIC-edge ide0 15: 22977880 0 03575943IO-APIC-edge ide1 16: 1526340728 1176101099 992391841 1707199189 IO-APIC-level eth0 17: 98016813 642505 961195082349025 IO-APIC-level eth1 NMI: 0 0 0 0 LOC: 1274300159 1274300179 1274300196 1274300195 ERR: 0 MIS: 0 My kernel is a # uname -ar Linux switch1 2.6.18-gentoo-r6 #1 SMP Wed Jan 24 21:08:48 CET 2007 i686 Intel(R) Xeon(TM) CPU 3.20GHz GenuineIntel GNU/Linux Tnx for attention Edoardo -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Edoardo Serra Envoyé : samedi 24 mars 2007 20:27 À : asterisk-users@lists.digium.com Objet : [asterisk-users] Re: RE : SIP/IAX peers UNREACHABLE and audio loss Martin Joseph ha scritto: The fact that qualify fails means you have a network issue. The same reason qualify fails (ie servers can't communicate) is the reason your users are experiencing quality issues in call. It was also my first though, but my LAN is very SIMPLE, so I was wondering if something else could cause the problem. turn off Qualify isn't going to fix anything IMO. It's just going to hide it from you. You're probably right, but it depends on Asterisk internals (which I don't know well). If Asterisk would stop to send RTP audio when just a qualify packet get lost it can make the situation worst. If the asterisk servers are all on your LAN then the network issue should be easily fixable. It should, but my LAN is very simple... I have a 10/100 Mbit switch with no more than 15 servers on it. Traffic on the LAN is not heavy even if the time of the day I see in the logs make me think it could be an issue related to network load trafic Anyhow I'll try to generate some heavy traffic on the LAN to see if it could be related to that. I also noticed that this problem began to happen when I upgraded my Asterisk to 1.2, but it can be a concidence. Do you think it could be related to bugs in ethernet drivers, kernel or whatever at the OS level ?? If the Asterisk servers are at remote locations and are using public internet, you might have problems resolving this completely. We have some Asterisk spread all over the public Internet, but firstly we should solve this problem at a LAN level Tnx for attention Regards Marty ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP/IAX peers UNREACHABLE and audio loss
Hi all, I'm having a problem with some Asterisk servers interconnected with each other using IAX (I also tried with SIP without solving the problem) Sometimes, with apparently no reason, some peers become UNREACHABLE (I have qualify=yes in iax.conf) and REACHABLE again as soon as another qualify test is made. Our users are also complaining about audio loss during their calls, apparently randomly, everything goes ok for days and bad for another few days. I strongly believe the 2 problems are strictly related because in the logs I see REACHABLE / UNREACHABLE messages only for certains days without regularity. The days in wich i see a lot of messages are exactly the days with most of complaint about audio loss I just noticed that timestamps of the logs (REACHABLE / UNREACHABLE) are quite always during business hours, this makes me think at somewhat related to load (cpu load, badwidth load, calls load, etc...) But, looking at hardware specs of our lan, servers and average load I don't think they are over-stressed. Our servers are all: 2 x Intel(R) Xeon(TM) CPU 3.20GHz 1 GB RAM 2 x IDE HDDs Software RAID 1 Asterisk 1.2.13 with res_perl Gentoo Linux Some of them has a Sangoma card connected with an E1 Most ot these are on the same LAN, interconnected with a 1 GB switch (I don't think it should be a bandwidth problem). Load averages of these server is varying from 0.5 to 1.0 (I guess it should be ok) On each server we don't have more than 50 concurrent calls (bridged SIP - IAX2 or IAX2 - ZAP) Used codec is mostly G729 Sometimes on asterisk cli i see some messages like Avoided initial deadlock for '0x9fd130', 10 retries! I don't know if it could be somehow related. Someone of you can point me in the right direction ? Tnx in advance Regards Ing. Edoardo Serra WeBRainstorm S.r.l. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Heartbeat on Digium T1 PCI cards?
Do you run asterisk through a wrapper as safe_asterisk ? (If not hi suggest you to do so) You can unload zaptel module from that script after a crash and reload it when the script tries to restart asterisk I'm using this solution on many production server whithout problems It sounds weird but I found it to be very useful with strange zaptel setup Hope it helps Regards Edoardo Shane Spencer ha scritto: I want to make sure that when an asterisk server dies that I am not left with a huge bill afterward for not hanging up a long distance call correctly. Are digium cards somehow set up to recieve a heartbeat from the drivers and if it skips a few beats it will take the t1 down in a way that would terminate the call? Shane ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk dropping audio
Hi all, I have a problem with Asterisk dropping audio. While in call, audio gets dropped for a while (from 5 to 60 secs, and obviously users often hangup, this means that I'm not sure the audio is always coming back after 60 secs), in the meantime the call remains up and no SIP signalation is generated. It happens randomly so it's very difficult to debug. I cannot see common circumstances when it happens (load average is always between 0.10 and 0.95, concurrent calls from 1 to 60 on a 2xXeon 3GHz with 2GB RAM). Calls are terminated to PSTN via other Asterisks with E1 (IAX2) or via SIP to other VoIP carriers. That problem happens with every different termination randomly, it also happens with calls between our users. (Well... I cannot exclude it's a termination problem, but I cannot find a common way to reproduce it) I'm using Asterisk 1.2.13 with res_perl (used to do lcr and to post customized cdr to mysql) I also tried 1.2.14 without solving that issue Kernel is a 2.6.18 vanilla on a linux gentoo I have g729 codec from digium installed and licensed, there are enough licenses available (I was tihinking of an issue of codec but I'm not sure it happens only with g729 calls) I now installed free g729 to see if it helps but I don't have any feedback yet I have an OpenSER acting as a load balancer for 2 asterisks but I don't think it could be responsible for that (I'm not using any kind of RTP proxy, rtp stream goes directly from user to asterisks) Every kind of help is really appreciated Regards Edoardo Serra WeBRainstorm S.r.l. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ISDN HFC card cannot 'detect remote answer'
Hi all, I have a box with an ISDN HFC card (1 BRI) connected to an Italian ISDN, the card is using zaphfc driver and it's receiving and originating calls quite regularly. There are some numbers (mostly toll free numbers) that I cannot connect to, here is what I get from the CLI: -- Called 5/803868 -- Zap/5-1 is proceeding passing it to SIP/184-11da8730 -- Zap/5-1 is making progress passing it to SIP/184-11da8730 But it never foes any further, it just hangs like that waiting for the timeout. My guess is the problem is related to the number being 'toll free', It seems to me that card isn't 'detecting a remote answer' and is not opening the audio channel. Does it make sense ? I'm new to ISDN protocol BTW, if I place the call with an FXO cards it works very well Here is my system configuration, hope it helps - Asterisk version is Asterisk 1.2.11-BRIstuffed-0.3.0-PRE-1s - Libpri version 1.2.3 - Zaptel version 1.2.8 - Linux distribution: Gentoo Tnx in advance for help Regards Edoardo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
