Re: [asterisk-users] Queue announcements and MOH blanking out on calls from PSTN over IAX2
On Monday 31 Oct 2011, Alex Kauffmann wrote: > Sorry if i missed it, but is IAX2 trunked? IF so, perhaps you are > running out of bandwidth in your IAX2 trunk. The setting > 'trunkmaxsize' defaults to 128000 bytes. > > From the readme file: > > "...Once this limit is > ; reached, calls may be dropped or begin to lose audio. Depending on > the codec in use and ; number of channels to be supported this value > may need to be raised, but in most cases the ; default value is > large enough." Normal calling (in and out) continues to work fine while this problem is being seen. We're not trunked, though -- each call sets up and tears down the connection. Just finished some preliminary testing on the same Asterisk on a 32-bit (instead of 64-bit) machine and the problem is not appearing. I'm reluctant to stamp this as the solution right now, since the test machine doesn't have any SIP peers at all (that's the only difference in the production and test beds), but am still hopeful. An i686 Asterisk installation on the production server should confirm or deny the theory that the problem was purely for Asterisk x64. Will post once we know more. Regards, -- Raj -- Raj Mathurr...@kandalaya.org http://kandalaya.org/ GPG: 78D4 FC67 367F 40E2 0DD5 0FEF C968 D0EF CC68 D17F PsyTrance & Chill: http://schizoid.in/ || It is the mind that moves -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] Queue announcements and MOH blanking out on calls from PSTN over IAX2
On 30/10/2011 05:53 a.m., Raj Mathur (राज माथुर) wrote: On Sunday 30 Oct 2011, Sammy Govind wrote: hmmm so IAX channel is playing with you guys. 1- Cant you guys use SIP, does this happen with SIP trunk as well !? 2- Which version of asterisk are there on both servers. 3- See the output of the command "core show file versions" in your both asterisk servers. Mainly lookout for IAX channel version. Also try enabling IAX debug and paste the output on console. 1.6.2.9-2+squeeze3 on the SIP server, 1.6.2.9-2+squeeze1 on the Dial server. I doubt if we'll be able to change the architecture of an infrastructure handling up to 450 simultaneous calls for the past 6 months at this stage, so SIP is out. IAX2 has been working beautifully for our needs up to this point, and we need to find a solution that we can integrate into this architecture itself! Incidentally, if anyone's interested, the installation itself is detailed at: http://www.mail-archive.com/ilugd@lists.linux-delhi.org/msg28166.html Regards, -- Raj Sorry if i missed it, but is IAX2 trunked? IF so, perhaps you are running out of bandwidth in your IAX2 trunk. The setting 'trunkmaxsize' defaults to 128000 bytes. From the readme file: "...Once this limit is ; reached, calls may be dropped or begin to lose audio. Depending on the codec in use and ; number of channels to be supported this value may need to be raised, but in most cases the ; default value is large enough." -- Alex -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] Queue announcements and MOH blanking out on calls from PSTN over IAX2
On Sun, Oct 30, 2011 at 12:04:09PM +0530, Raj Mathur (राज माथुर) wrote: > On Sunday 30 Oct 2011, Raj Mathur (राज माथुर) wrote: > > After looking further, the problem seems to be purely in playing > > recorded messages over IAX2. Looking at the debug logs on the SIP > > server (which is playing the recorded messages) shows that it stops > > playing one of the messages at some point in the flow, and then never > > plays anything again. > > This seems to be very similar to: > > https://issues.asterisk.org/view.php?id=17232 > > except there is no virtualisation involved in the process -- everything > is working on native hardware. It /is/ amd64 Debian Squeeze running on > Intel, though. Do you use DAHDI timing? Try 'timing test' in the Asterisk CLI. If so, I wonder if it's possible that the redfone devices may actually have occasional hiccups as a timing source[1]. This should be easily noticable using 'dahdi_test'. Anyway, maybe try a different timing source, by disabling other res_timing*.so modules in modules.conf (and restarting asterisk). [1] Sorry, I'm not familiar with them well enough, and apologize in advance if this suggestion is silly. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue announcements and MOH blanking out on calls from PSTN over IAX2
On Sunday 30 Oct 2011, Sammy Govind wrote: > hmmm so IAX channel is playing with you guys. > > 1- Cant you guys use SIP, does this happen with SIP trunk as well !? > 2- Which version of asterisk are there on both servers. > 3- See the output of the command "core show file versions" in your > both asterisk servers. Mainly lookout for IAX channel version. > > Also try enabling IAX debug and paste the output on console. 1.6.2.9-2+squeeze3 on the SIP server, 1.6.2.9-2+squeeze1 on the Dial server. I doubt if we'll be able to change the architecture of an infrastructure handling up to 450 simultaneous calls for the past 6 months at this stage, so SIP is out. IAX2 has been working beautifully for our needs up to this point, and we need to find a solution that we can integrate into this architecture itself! Incidentally, if anyone's interested, the installation itself is detailed at: http://www.mail-archive.com/ilugd@lists.linux-delhi.org/msg28166.html Regards, -- Raj -- Raj Mathurr...@kandalaya.org http://kandalaya.org/ GPG: 78D4 FC67 367F 40E2 0DD5 0FEF C968 D0EF CC68 D17F PsyTrance & Chill: http://schizoid.in/ || It is the mind that moves -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] Queue announcements and MOH blanking out on calls from PSTN over IAX2
hmmm so IAX channel is playing with you guys. 1- Cant you guys use SIP, does this happen with SIP trunk as well !? 2- Which version of asterisk are there on both servers. 3- See the output of the command "core show file versions" in your both asterisk servers. Mainly lookout for IAX channel version. Also try enabling IAX debug and paste the output on console. 2011/10/30 Raj Mathur (राज माथुर) > On Sunday 30 Oct 2011, Raj Mathur (राज माथुर) wrote: > > After looking further, the problem seems to be purely in playing > > recorded messages over IAX2. Looking at the debug logs on the SIP > > server (which is playing the recorded messages) shows that it stops > > playing one of the messages at some point in the flow, and then never > > plays anything again. > > This seems to be very similar to: > > https://issues.asterisk.org/view.php?id=17232 > > except there is no virtualisation involved in the process -- everything > is working on native hardware. It /is/ amd64 Debian Squeeze running on > Intel, though. > > Regards, > > -- Raj > -- > Raj Mathurr...@kandalaya.org http://kandalaya.org/ > GPG: 78D4 FC67 367F 40E2 0DD5 0FEF C968 D0EF CC68 D17F > PsyTrance & Chill: http://schizoid.in/ || It is the mind that moves > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every 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] Queue announcements and MOH blanking out on calls from PSTN over IAX2
On Sunday 30 Oct 2011, Raj Mathur (राज माथुर) wrote: > After looking further, the problem seems to be purely in playing > recorded messages over IAX2. Looking at the debug logs on the SIP > server (which is playing the recorded messages) shows that it stops > playing one of the messages at some point in the flow, and then never > plays anything again. This seems to be very similar to: https://issues.asterisk.org/view.php?id=17232 except there is no virtualisation involved in the process -- everything is working on native hardware. It /is/ amd64 Debian Squeeze running on Intel, though. Regards, -- Raj -- Raj Mathurr...@kandalaya.org http://kandalaya.org/ GPG: 78D4 FC67 367F 40E2 0DD5 0FEF C968 D0EF CC68 D17F PsyTrance & Chill: http://schizoid.in/ || It is the mind that moves -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] Queue announcements and MOH blanking out on calls from PSTN over IAX2
On Sunday 30 Oct 2011, Sammy Govind wrote:
> Try turning on the Sip debug for the PSTN call as well as RTP debug.
> Paste the output here.
There's no SIP involved until the call is picked up -- only PSTN and
IAX2. No RTP log available either.
> > The Dial server is connected to multiple 4-port Redfone devices for
> > handling PSTN incoming and outgoing calls. Outgoing calls always
> > originate from and incoming calls always terminate at the SIP
> > server. SIP and Dial servers are connected over IAX2.
>
> Explain the above abit as well..couldnt get the clear picture of what
> it looks like. Seems to me that you guys are using two servers and
> call-audio gets lost in between the servers OR in between the
> Dial-Server and redfone device for Queue Calls.
After looking further, the problem seems to be purely in playing
recorded messages over IAX2. Looking at the debug logs on the SIP
server (which is playing the recorded messages) shows that it stops
playing one of the messages at some point in the flow, and then never
plays anything again.
Here's the logs from three different calls:
*** Announcement gets cut off in the "please hold" message
> requested format = alaw,
> requested prefs = (alaw|gsm|ulaw),
> actual format = alaw,
> host prefs = (alaw),
> priority = mine
-- Executing [6000@cg-20:19] NoOp("IAX2/dialbank-1-5531", "Spygroup done")
in new
stack
-- Executing [6000@cg-20:20] Answer("IAX2/dialbank-1-5531", "") in new stack
-- Executing [6000@cg-20:21] Wait("IAX2/dialbank-1-5531", "1") in new stack
-- Executing [6000@cg-20:22] BackGround("IAX2/dialbank-1-5531",
"followme/pls-hold-
while-try") in new stack
-- Playing 'followme/pls-hold-while-try.slin'
(language 'en')
*** Caller hung up after some time of silence
== Spawn extension (cg-20, 6000, 22) exited non-zero on 'IAX2/dialbank-1-5531'
-- Hungup 'IAX2/dialbank-1-5531'
*** Announcement gets cut off in the first
*** "Your call is now next in line" message
-- Accepting AUTHENTICATED call from 10.0.10.132:
> requested format = alaw,
> requested prefs = (alaw|gsm|ulaw),
> actual format = alaw,
> host prefs = (alaw),
> priority = mine
-- Executing [6000@cg-20:20] Answer("IAX2/dialbank-1-4966", "") in new stack
-- Executing [6000@cg-20:21] Wait("IAX2/dialbank-1-4966", "1") in new stack
-- Executing [6000@cg-20:22] BackGround("IAX2/dialbank-1-4966",
"followme/pls-hold-
while-try") in new stack
-- Playing 'followme/pls-hold-while-try.slin'
(language 'en')
-- Executing [6000@cg-20:23] Wait("IAX2/dialbank-1-4966", "1") in new stack
-- Executing [6000@cg-20:32] Queue("IAX2/dialbank-1-4966",
"cg-20,dr") in new
stack
-- Playing 'queue-youarenext.slin' (language 'en')
*** Caller hung up after some time of silence
== Spawn extension (cg-20, 6000, 32) exited non-zero on 'IAX2/dialbank-1-4966'
-- Hungup 'IAX2/dialbank-1-4966'
*** Caller hears "please hold", "Your call is now next in line",
*** "thank you for your patience" messages, a few rings,
*** then sound gets cut off in the next "Your call is now next in line"
*** message
-- Accepting AUTHENTICATED call from 10.0.10.132:
> requested format = alaw,
> requested prefs = (alaw|gsm|ulaw),
> actual format = alaw,
> host prefs = (alaw),
> priority = mine
-- Executing [6000@cg-20:20] Answer("IAX2/dialbank-1-2655", "") in new stack
-- Executing [6000@cg-20:21] Wait("IAX2/dialbank-1-2655", "1") in new stack
-- Executing [6000@cg-20:22] BackGround("IAX2/dialbank-1-2655",
"followme/pls-hold-
while-try") in new stack
-- Playing 'followme/pls-hold-while-try.slin'
(language 'en')
-- Executing [6000@cg-20:23] Wait("IAX2/d
Re: [asterisk-users] Queue announcements and MOH blanking out on calls from PSTN over IAX2
Try turning on the Sip debug for the PSTN call as well as RTP debug. Paste the output here. > > The Dial server is connected to multiple 4-port Redfone devices for > handling PSTN incoming and outgoing calls. Outgoing calls always > originate from and incoming calls always terminate at the SIP server. > SIP and Dial servers are connected over IAX2. Explain the above abit as well..couldnt get the clear picture of what it looks like. Seems to me that you guys are using two servers and call-audio gets lost in between the servers OR in between the Dial-Server and redfone device for Queue Calls. 2011/10/29 Raj Mathur (राज माथुर) > On Saturday 29 Oct 2011, Raj Mathur (राज माथुर) wrote: > > [snip] > > Callers coming in from the PSTN (through the Dial server, over IAX2) > > can also talk normally after an agent has picked up the call. > > However, callers from the PSTN get the announcement and/or MOH > > blanked out after a random period of time, typically 5-10 seconds. > > This often happens in the middle of the queue position or thank-you > > announcement. > > > > After the blanking out, the call is still alive, queue functions are > > working, and if an agent picks up the calls s/he can talk normally to > > the caller. However, blanking out of the MOH/announcement makes the > > caller think that the call has been dropped, and they hang up before > > an agent answers. > > > > Debug logs show that Asterisk is playing the MOH and announcement > > files continuously even though the caller cannot hear them. > > > > Unable to figure out why the blanking happens ONLY on incoming calls > > from the PSTN. Any help appreciated. > > Further simplified the issue to an extension that just does: > > ... Answer() > ... MusicOnHold(default) > > When called from the PSTN, the musiconhold blanks out after a few > seconds, while it plays fine when the extension is called locally. > > Regards, > > -- Raj > -- > Raj Mathurr...@kandalaya.org http://kandalaya.org/ > GPG: 78D4 FC67 367F 40E2 0DD5 0FEF C968 D0EF CC68 D17F > PsyTrance & Chill: http://schizoid.in/ || It is the mind that moves > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every 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] Queue announcements and MOH blanking out on calls from PSTN over IAX2
On Saturday 29 Oct 2011, Raj Mathur (राज माथुर) wrote: > [snip] > Callers coming in from the PSTN (through the Dial server, over IAX2) > can also talk normally after an agent has picked up the call. > However, callers from the PSTN get the announcement and/or MOH > blanked out after a random period of time, typically 5-10 seconds. > This often happens in the middle of the queue position or thank-you > announcement. > > After the blanking out, the call is still alive, queue functions are > working, and if an agent picks up the calls s/he can talk normally to > the caller. However, blanking out of the MOH/announcement makes the > caller think that the call has been dropped, and they hang up before > an agent answers. > > Debug logs show that Asterisk is playing the MOH and announcement > files continuously even though the caller cannot hear them. > > Unable to figure out why the blanking happens ONLY on incoming calls > from the PSTN. Any help appreciated. Further simplified the issue to an extension that just does: ... Answer() ... MusicOnHold(default) When called from the PSTN, the musiconhold blanks out after a few seconds, while it plays fine when the extension is called locally. Regards, -- Raj -- Raj Mathurr...@kandalaya.org http://kandalaya.org/ GPG: 78D4 FC67 367F 40E2 0DD5 0FEF C968 D0EF CC68 D17F PsyTrance & Chill: http://schizoid.in/ || It is the mind that moves -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Queue announcements and MOH blanking out on calls from PSTN over IAX2
Hi, Problem with Asterisk 1.6.2.9 on Debian Squeeze. * Infrastructure We have two servers, SIP and Dial. The SIP server handles SIP clients; it also receives incoming PSTN calls from the Dial server and makes outgoing PSTN calls on the Dial server. The Dial server is connected to multiple 4-port Redfone devices for handling PSTN incoming and outgoing calls. Outgoing calls always originate from and incoming calls always terminate at the SIP server. SIP and Dial servers are connected over IAX2. Normal incoming and outgoing have been working fine for many months now on this setup. * Problem We recently enabled caller queues on the SIP server. Queue functions are working fine. Local (on SIP server itself) callers get periodic position announcements and MOH while they wait for the call to be picked up. Once an agent picks up the call, the caller and agent can talk normally. Tried with an IAX2 client (instead of SIP) and that works fine too. Callers coming in from the PSTN (through the Dial server, over IAX2) can also talk normally after an agent has picked up the call. However, callers from the PSTN get the announcement and/or MOH blanked out after a random period of time, typically 5-10 seconds. This often happens in the middle of the queue position or thank-you announcement. After the blanking out, the call is still alive, queue functions are working, and if an agent picks up the calls s/he can talk normally to the caller. However, blanking out of the MOH/announcement makes the caller think that the call has been dropped, and they hang up before an agent answers. Debug logs show that Asterisk is playing the MOH and announcement files continuously even though the caller cannot hear them. Unable to figure out why the blanking happens ONLY on incoming calls from the PSTN. Any help appreciated. Regards, -- Raj -- Raj Mathurr...@kandalaya.org http://kandalaya.org/ GPG: 78D4 FC67 367F 40E2 0DD5 0FEF C968 D0EF CC68 D17F PsyTrance & Chill: http://schizoid.in/ || It is the mind that moves -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
