[asterisk-users] Automatic jump from line to line for incoming calls and the problem in DAHDI
Dears; I am facing the following problem: Already we requested from the service provider to enable the auto jumping service for our analoge telephone lines, so because we have 4 telephone lines from the service provider, then if you called line # 1 and it was busy, then the call will be sent to any available line #2 or #3 or #4, and if you call line # 3 and it was busy then the call will be sent via any available line of these four lines. This feature is causing a problem at the Asterisk PBX, so some calls are not handled properly (it is ringing and we do not hear the welcome message), also the outgoing calls are facing a problem because it seems that there is a confusing happening in dahdi to determine the available line. I do not know really how the automatic jumping feature is working at the service provider and what is the effecting on the DAHDI and Asterisk that is causing to not responding for the DAHDI channels properly. For more details to be sure that I described the behaviour of the auto jumping feature that I took it from the service provider, let us assume my number is 22446789, when I call this number and I look for asterisk CLI, I can see that the call came via DAHDI/3-1 and then I do another call to this line, I can see it via DAHDI/4-1 and I do another call to this line and I will see it via DAHDI/2-1. Also, not all my calls are failed ... but some are succeed and some are fails, so the responding is not perfect. I am sure because of the auto jumping feature from the service provider. Appreciate the kindly help and advise. Regards Bilal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Automatic jump from line to line for incoming calls and the problem in DAHDI
On Wednesday 17 October 2012, bilal ghayyad wrote:
Dears;
I am facing the following problem:
Already we requested from the service provider to enable the auto jumping
service for our analoge telephone lines, so because we have 4 telephone
lines from the service provider, then if you called line # 1 and it was
busy, then the call will be sent to any available line #2 or #3 or #4, and
if you call line # 3 and it was busy then the call will be sent via any
available line of these four lines.
This feature is causing a problem at the Asterisk PBX, so some calls are
not handled properly (it is ringing and we do not hear the welcome
message), also the outgoing calls are facing a problem because it seems
that there is a confusing happening in dahdi to determine the available
line.
I do not know really how the automatic jumping feature is working at the
service provider and what is the effecting on the DAHDI and Asterisk that
is causing to not responding for the DAHDI channels properly.
For more details to be sure that I described the behaviour of the auto
jumping feature that I took it from the service provider, let us assume my
number is 22446789, when I call this number and I look for asterisk CLI, I
can see that the call came via DAHDI/3-1 and then I do another call to
this line, I can see it via DAHDI/4-1 and I do another call to this line
and I will see it via DAHDI/2-1.
Also, not all my calls are failed ... but some are succeed and some are
fails, so the responding is not perfect. I am sure because of the auto
jumping feature from the service provider.
If you have multiple lines, and they are all paid for in the same name, then
your telco really should have set it up so they are all accessible by dialling
the same number.
Way back in the clicky-clicky days, having multiple lines connected to the
same switchboard would have been done at the exchange by allocating sequential
lines on the same selector, which was modified to step on until it found a non-
engaged line (or go to engaged tone, if the last in the set were engaged).
For instance, Radio Derby's main switchboard number was 36; but 361112,
361113, 361114, 361115 or 361116 might also reach the switchboard (depending
whether or not that line was already in use).
Digital exchanges don't have such requirements, of course; and since we went
over to System X, which does not impose a 1:1 mapping between (logical)
numbers and (physical) lines, 361112 (at least) has been allocated to
another subscriber. And there are many lines numbered 36.
If you have several lines and they are properly grouped by the telco, you may
get a call coming in via a differently-numbered line than what the other
subscriber actually dialled.
The way top deal with this in Asterisk is as follows: Have one context that
handles incoming calls from the PSTN (usually [from-pstn] but you may have
changed this). In this context, you just need to handle calls for any
extension the same. (Or make sure, by using a catch-all such as the s
extension or _X.)
For calling out, make sure all your DAHDI channels are in the same group in
chan_dahdi.conf, and use something in your Dial() command like
Dial(DAHDI/g1/${EXTEN}) or Dial(DAHDI/r1/${EXTEN}) . The g form will try
always to use the lowest-numbered available channel; the r form will keep a
track of which channel was used last and try to cycle through channels in turn
from lowest to highest. (Capital G1 and R1 will try always to use the highest
number, and cycle through from high to low respectively).
--
AJS
Answers come *after* questions.
--
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[asterisk-users] Fully utilise all PRIs in a DAHDI group
Hi, Our client has DAHDI groups with 4 PRIs in each group (one 4-port interface per group), up to 6 groups per server. When we dial, we can specify the group to be used for dialling, and our dial plan automatically distributes calls over multiple servers and multiple groups within a server. The way Asterisk dials by default is to use the lowest-numbered free line in a group to place a call. This is technically fine. However, what it means for our client is that the first couple of PRIs in a group tend to get the bulk of calls, the other two remain more-or-less unutilised. This is a problem, since there are call commitments to the Telco for each PRI line. The Telco tends to get all soggy and hard to light if some of the PRIs are used way below committed call levels. One solution is to group at the individual PRI level, so the load balancing automatically takes care of fair utilisation of each PRI. However, for various reasons we'd prefer not to do this. Another solution would be if Asterisk could choose a random (or LRU or LCU or round-robin or any other scheme) PRI within a group when dialling. Any roughly fair way to distribute calls to PRIs within a DAHDI group would be fine. Is there some way to achieve this? Asterisk 1.8.8 on Debian Squeeze. Regards, -- Raj -- Raj Mathur || r...@kandalaya.org || GPG: http://otheronepercent.blogspot.com || http://kandalaya.org || CC68 It is the mind that moves || http://schizoid.in || D17F -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] question on softhangup
Dave Platt provided the following answer to a similar question of mine last week. I was trying to use SoftHangup() to prempt a DAHDI line for an emergency call. Here is his reply. That may be due to a common characteristic of PSTN lines (at least, it's common here in the U.S.) By design, most U.S. PSTN lines have a very asymmetrical response to a physical hangup: - If the calling party hangs up, the call is terminated immediately. - If the called party hangs up, and the calling party does not, the line remains live for some time (typically around 30 seconds, I believe). If the called party goes off-hook again during this period, they can resume the call. If I recall correctly, things were designed this way so that the called party could say Oh, hang on, I answered this call in the bedroom and the stuff I need is in the living room, hang up the extension phone, go to another room, pick up the other phone and carry on with the call. If that's what you're running into here - if the line you are trying to SoftHangup() was handing an inbound call - then there may be no good solution. As far as I know, there is no way to force an incoming PSTN call to release the line, other than go on-hook, and wait for 30 seconds to pass. Several possible workarounds, roughly in order of increasing complexity and decreasing reliability: (1) Keep one of your PSTN lines reserved for emergency calls only; remove it from your inbound hunt group and place it in a Dahdi line group of its own (or don't group it at all). (2) Keep one of your PSTN lines reserved for *outbound* calls only; you should be able to SoftHangup() an outbound call within a second or two. (3) Figure out a way to check the PSTN lines that are in use at the time of an emergency - if they're all in use, somehow find one which was in use for an outbound call, and select it as the one to SoftHangup() and dial upon. (4) If you must keep all of your PSTN lines in bidirectional use, you may have to *tell* the parties that the line is needed for an emergency call, and ask them to release the line. Do a barge-in on the channel, play an alert sound, play a message saying Emergency call in progress, please hang up this line immediately, play the alert sound again for a few seconds, SoftHangup(), Wait(2), and then try dialing. Mitch On 10/16/2012 08:59 PM, Jerry Geis wrote: How do I use softhangup through the AMI interface? I am using 1.4.43. Will softhangup hangup a DAHDI channel? I have found that Action: Hangup does not hangup a DAHDI channel only sip. Thanks, jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] Fully utilise all PRIs in a DAHDI group
In article 201210171813.45334.r...@linux-delhi.org, Raj Mathur (राठमाथॠर) r...@linux-delhi.org wrote: Hi, Our client has DAHDI groups with 4 PRIs in each group (one 4-port interface per group), up to 6 groups per server. When we dial, we can specify the group to be used for dialling, and our dial plan automatically distributes calls over multiple servers and multiple groups within a server. The way Asterisk dials by default is to use the lowest-numbered free line in a group to place a call. This is technically fine. However, what it means for our client is that the first couple of PRIs in a group tend to get the bulk of calls, the other two remain more-or-less unutilised. This is a problem, since there are call commitments to the Telco for each PRI line. The Telco tends to get all soggy and hard to light if some of the PRIs are used way below committed call levels. One solution is to group at the individual PRI level, so the load balancing automatically takes care of fair utilisation of each PRI. However, for various reasons we'd prefer not to do this. Another solution would be if Asterisk could choose a random (or LRU or LCU or round-robin or any other scheme) PRI within a group when dialling. Any roughly fair way to distribute calls to PRIs within a DAHDI group would be fine. Is there some way to achieve this? Asterisk 1.8.8 on Debian Squeeze. Instead of dialling using DAHDI/g1/123456789, you can try using DAHDI/r1/123456789 to make Asterisk use the channels in round-robin order instead of always choosing the lowest free channel. See http://www.voip-info.org/wiki/view/Asterisk+ZAP+channels (I could not find comparable information on the Asterisk WIKI at https://wiki.asterisk.org). Cheers Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fully utilise all PRIs in a DAHDI group
On Wed, Oct 17, 2012 at 8:43 AM, Raj Mathur (राज माथुर) r...@linux-delhi.org wrote: Hi, Our client has DAHDI groups with 4 PRIs in each group (one 4-port interface per group), up to 6 groups per server. When we dial, we can specify the group to be used for dialling, and our dial plan automatically distributes calls over multiple servers and multiple groups within a server. The way Asterisk dials by default is to use the lowest-numbered free line in a group to place a call. This is technically fine. However, what it means for our client is that the first couple of PRIs in a group tend to get the bulk of calls, the other two remain more-or-less unutilised. This is a problem, since there are call commitments to the Telco for each PRI line. The Telco tends to get all soggy and hard to light if some of the PRIs are used way below committed call levels. One solution is to group at the individual PRI level, so the load balancing automatically takes care of fair utilisation of each PRI. However, for various reasons we'd prefer not to do this. Another solution would be if Asterisk could choose a random (or LRU or LCU or round-robin or any other scheme) PRI within a group when dialling. Any roughly fair way to distribute calls to PRIs within a DAHDI group would be fine. Is there some way to achieve this? Asterisk 1.8.8 on Debian Squeeze. Regards, -- Raj -- Raj Mathur || r...@kandalaya.org || GPG: http://otheronepercent.blogspot.com || http://kandalaya.org || CC68 It is the mind that moves || http://schizoid.in || D17F Taken from the wiki searching with the exact terms you used. http://www.voip-info.org/wiki/view/Asterisk+ZAP+channels Thanks, Steve Totaro Dialing a GroupIn the Zap Channel Module's configuration file (zapata.confhttp://www.voip-info.org/wiki/view/Asterisk+config+zapata.conf), you can define groups of Zap channels that get treated as a single channel as far as the Dial http://www.voip-info.org/wiki/view/Asterisk+cmd+Dialcommand is concerned. You specify which of four methods the Zap channel module is to use to select a non-busy channel from the channel group by prefixing the group number with one of the letters *g*, *G*, *r*, or *R*: - *g*: select the lowest-numbered non-busy Zap channel (aka. ascending sequential hunt group). - *G*: select the highest-numbered non-busy Zap channel (aka. descending sequential hunt group). - *r*: use a round-robin search, starting at the next highest channel than last time (aka. ascending rotary hunt group). - *R*: use a round-robin search, starting at the next lowest channel than last time (aka. descending rotary hunt group). The round-robin searches make the Zap channel module start looking for an available channel from a different channel number each time. For each channel group, the Zap channel module keeps track of the last round-robin start point, and this time starts checking availability from either the next (lowercase *r*)) or the previous uppercase *R* channel in the group. Which channel it actually finds available (if any) does not affect the starting point for the next round-robin search. Calls to the Dialhttp://www.voip-info.org/wiki/view/Asterisk+cmd+Dial command using ordinary (*g* or *G*) group selections do not affect future round-robin starting points either. For example, if you have defined channel group 2 as containing Zap channels 1, 2, 5 and 8, and the last round-robin search for this group (group 2) began searching from channel 5, this is the order of searching that the Zap channel module will use for the four possible selection methods: - Dial(Zap/g2...): Looks in order 1, 2, 5, 8 - Dial(Zap/G2...): Looks in order 8, 5, 2, 1 - Dial(Zap/r2...): Looks in order 8, 1, 2, 5 - Dial(Zap/R2...): Looks in order 2, 1, 8, 5 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.8.17.0 Requested transfer capability: 0x00 - SPEECH
Hello, I posted this problems in the past and was not able to find the solution, at the time I posted this issue I had a equipment malfunction which has been fixed but I still have the -- Requested transfer capability: 0x00 - SPEECH error. Any suggestions? Asterisk 1.8.17.0 built by root @ as2.x.com on a x86_64 running Linux on 2012-10-16 22:38:14 UTC == Using SIP RTP CoS mark 5 -- Executing [2663636@voipphones:1] Set(SIP/4856-, CALLERID(num)=2663636) in new stack -- Executing [2663636@voipphones:2] Dial(SIP/4856-, dahdi/g1/2663636) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called dahdi/g1/2663636 -- DAHDI/i1/2663636-1 is proceeding passing it to SIP/4856- -- DAHDI/i1/2663636-1 is ringing -- DAHDI/i1/2663636-1 is making progress passing it to SIP/4856- -- Hungup 'DAHDI/i1/2663636-1' == Spawn extension (voipphones, 2663636, 2) exited non-zero on 'SIP/4856-' Thanks in Advance -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8.17.0 Requested transfer capability: 0x00 - SPEECH
motty.cruz wrote: Hello, Hola, I posted this problems in the past and was not able to find the solution, at the time I posted this issue I had a equipment malfunction which has been fixed but I still have the -- Requested transfer capability: 0x00 - SPEECH error. That's not an error. Are you experiencing some issue that is causing you to think it is? Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fully utilise all PRIs in a DAHDI group
On Wednesday 17 Oct 2012, Tony Mountifield wrote: In article 201210171813.45334.r...@linux-delhi.org, Raj Mathur (à€°à€Ÿà€ à€®à€Ÿà€¥à¥ à€°) r...@linux-delhi.org wrote: Our client has DAHDI groups with 4 PRIs in each group (one 4-port interface per group), up to 6 groups per server. When we dial, we can specify the group to be used for dialling, and our dial plan automatically distributes calls over multiple servers and multiple groups within a server. The way Asterisk dials by default is to use the lowest-numbered free line in a group to place a call. This is technically fine. However, what it means for our client is that the first couple of PRIs in a group tend to get the bulk of calls, the other two remain more-or-less unutilised. This is a problem, since there are call commitments to the Telco for each PRI line. The Telco tends to get all soggy and hard to light if some of the PRIs are used way below committed call levels. One solution is to group at the individual PRI level, so the load balancing automatically takes care of fair utilisation of each PRI. However, for various reasons we'd prefer not to do this. Another solution would be if Asterisk could choose a random (or LRU or LCU or round-robin or any other scheme) PRI within a group when dialling. Any roughly fair way to distribute calls to PRIs within a DAHDI group would be fine. Is there some way to achieve this? Asterisk 1.8.8 on Debian Squeeze. Instead of dialling using DAHDI/g1/123456789, you can try using DAHDI/r1/123456789 to make Asterisk use the channels in round-robin order instead of always choosing the lowest free channel. See http://www.voip-info.org/wiki/view/Asterisk+ZAP+channels (I could not find comparable information on the Asterisk WIKI at https://wiki.asterisk.org). Thanks to you and Steve Totaro, that's exactly what I need. Regards, -- Raj -- Raj Mathur || r...@kandalaya.org || GPG: http://otheronepercent.blogspot.com || http://kandalaya.org || CC68 It is the mind that moves || http://schizoid.in || D17F -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8.17.0 Requested transfer capability: 0x00 - SPEECH
Thanks Joshua, Actually you're right, I'm not experiencing any issue. I don't see that Requested transfer capability: 0x00 - SPEECH in older version of Asterisk so I was wondering? And since my is from PRI to BPX. Thanks -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joshua Colp Sent: Wednesday, October 17, 2012 7:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 1.8.17.0 Requested transfer capability: 0x00 - SPEECH motty.cruz wrote: Hello, Hola, I posted this problems in the past and was not able to find the solution, at the time I posted this issue I had a equipment malfunction which has been fixed but I still have the -- Requested transfer capability: 0x00 - SPEECH error. That's not an error. Are you experiencing some issue that is causing you to think it is? Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8.17.0 Requested transfer capability: 0x00 - SPEECH
motty.cruz wrote: Thanks Joshua, Actually you're right, I'm not experiencing any issue. I don't see that Requested transfer capability: 0x00 - SPEECH in older version of Asterisk so I was wondering? And since my is from PRI to BPX. The message is only printed out if your verbose level is at a certain number or higher. It's nothing to be worried about. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Motif/XMPP for Google Voice
I again would recommend a more thorough explanation of the configs I've been using Asterisk for years - but the configs for this need some explanation in the wiki The samples contradict what the wiki has.. And as I indicated I could not get audio working... On 10/15/12 10:11 AM, Joshua Colp jc...@digium.com wrote: Joshua Colp wrote: asterisk asterisk wrote: Dear all, Hola, I wish to ask a question of the new Motif Channel in asterisk 11. I successfully compile the binary and run without error. However, when dialing out, no external connection only ringing. During testing some issues were uncovered with the Motif channel driver, but unfortunately they did not make the last release candidate. My suggestion is to get Asterisk 11 from SVN or if you are not comfortable with that wait until the official Asterisk 11 release. The fixes did not make the last release candidate, that is. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Motif/XMPP for Google Voice
Robert wrote: I again would recommend a more thorough explanation of the configsŠ The sample configuration details the various options of the channel driver and some very simple generic examples, they aren't made to just work for various services and clients. I've been using Asterisk for years - but the configs for this need some explanation in the wikiŠ How can they be made better in your opinion? What do you feel is lacking and want to see? I feel as though what you are asking for isn't more explanation of the configuration but more details on how to configure it for use with *specific* services and clients. The sample tells you what the various options means and how things work but the combination of them ultimately depends on that. I would be fine with such things being written. The samples contradict what the wiki has.. And as I indicated I could not get audio working... The wiki entry available at https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google details using chan_motif with Google Voice in more detail. Can you explain how it contradicts the samples in your opinion? As for you not getting audio working - issues were discovered that have been fixed. These fixes will be in the next release candidate. We also changed a default setting to no for ICE support, this needs to be set to yes for chan_motif to operate and I have updated the wiki to reflect this. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Automatic jump from line to line for incoming calls and the problem in DAHDI
Actually I am not talking on how to handle it in the extensions.conf because I
am doing same as you wrote. But even so, I am facing a problem that some calls
are captured and some calls are not captured.
Currently, I set the callwaiting=no in the chan_dahdi.conf, it seems it is
working fine. But I am not sure if this is really the required configuration to
fix it or there is something else.
Any advise.
Regards
Bilal
Dears;
I am facing the following problem:
Already we requested from the service provider to
enable the auto jumping
service for our analoge telephone lines, so because we
have 4 telephone
lines from the service provider, then if you called
line # 1 and it was
busy, then the call will be sent to any available line
#2 or #3 or #4, and
if you call line # 3 and it was busy then the call will
be sent via any
available line of these four lines.
This feature is causing a problem at the Asterisk PBX,
so some calls are
not handled properly (it is ringing and we do not hear
the welcome
message), also the outgoing calls are facing a problem
because it seems
that there is a confusing happening in dahdi to
determine the available
line.
I do not know really how the automatic jumping feature
is working at the
service provider and what is the effecting on the DAHDI
and Asterisk that
is causing to not responding for the DAHDI channels
properly.
For more details to be sure that I described the
behaviour of the auto
jumping feature that I took it from the service
provider, let us assume my
number is 22446789, when I call this number and I look
for asterisk CLI, I
can see that the call came via DAHDI/3-1 and then I do
another call to
this line, I can see it via DAHDI/4-1 and I do another
call to this line
and I will see it via DAHDI/2-1.
Also, not all my calls are failed ... but some are
succeed and some are
fails, so the responding is not perfect. I am sure
because of the auto
jumping feature from the service provider.
If you have multiple lines, and they are all paid for in the
same name, then
your telco really should have set it up so they are all
accessible by dialling
the same number.
Way back in the clicky-clicky days, having multiple lines
connected to the
same switchboard would have been done at the exchange by
allocating sequential
lines on the same selector, which was modified to step on
until it found a non-
engaged line (or go to engaged tone, if the last in
the set were engaged).
For instance, Radio Derby's main switchboard number was
36; but 361112,
361113, 361114, 361115 or 361116 might also reach the
switchboard (depending
whether or not that line was already in use).
Digital exchanges don't have such requirements, of course;
and since we went
over to System X, which does not impose a 1:1 mapping
between (logical)
numbers and (physical) lines, 361112 (at
least) has been allocated to
another subscriber. And there are many lines numbered
36.
If you have several lines and they are properly grouped by
the telco, you may
get a call coming in via a differently-numbered line than
what the other
subscriber actually dialled.
The way top deal with this in Asterisk is as follows:
Have one context that
handles incoming calls from the PSTN (usually
[from-pstn] but you may have
changed this). In this context, you just need to
handle calls for any
extension the same. (Or make sure, by using a
catch-all such as the s
extension or _X.)
For calling out, make sure all your DAHDI channels are in
the same group in
chan_dahdi.conf, and use something in your Dial() command
like
Dial(DAHDI/g1/${EXTEN}) or Dial(DAHDI/r1/${EXTEN}) .
The g form will try
always to use the lowest-numbered available channel; the r
form will keep a
track of which channel was used last and try to cycle
through channels in turn
from lowest to highest. (Capital G1 and R1 will try
always to use the highest
number, and cycle through from high to low respectively).
--
AJS
Answers come *after* questions.
--
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[asterisk-users] Agents in more than one queue at once
My company has been running Asterisk 1.6.2.19-1_centos5 from the official yum repo, and for a while now I've been receiving complaints from our call centers about calls not being routed in the most efficient order. I'll explain with a simplified scenario-- Let's say I have two queues: A and B. I have one agent, Alice, who is a member of both of these queues. While Alice is busy on a call, one person calls in to queue A, and then, several moments later, another person calls in to queue B. At this point, note that both callers waiting on hold are position 1 in their respective queues. A queue show might look like this... A has 1 calls (max unlimited) in 'leastrecent' strategy (0s holdtime, 533s talktime), W:1, C:1, A:0, SL:100.0% within 60s Members: 21 (Local/21@from-queue/n) (dynamic) (In use) has taken 1 calls (last was 533 secs ago) Callers: 1. SIP/Trunk-eb17 (wait: 1:14, prio: 0) B has 1 calls (max unlimited) in 'leastrecent' strategy (0s holdtime, 533s talktime), W:1, C:1, A:0, SL:100.0% within 60s Members: 21 (Local/21@from-queue/n) (dynamic) (In use) has taken 1 calls (last was 533 secs ago) Callers: 1. SIP/Trunk-eb1e (wait: 0:45, prio: 0) My question is: when Alice gets off the phone, which call will she get? My expectation is that she will get the call which has been waiting longer, but I'm not sure that's actually the case. Alex Forster -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] Agents in more than one queue at once
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex Forster Sent: Wednesday, October 17, 2012 2:42 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Agents in more than one queue at once My company has been running Asterisk 1.6.2.19-1_centos5 from the official yum repo, and for a while now I've been receiving complaints from our call centers about calls not being routed in the most efficient order. I'll explain with a simplified scenario-- Let's say I have two queues: A and B. I have one agent, Alice, who is a member of both of these queues. While Alice is busy on a call, one person calls in to queue A, and then, several moments later, another person calls in to queue B. At this point, note that both callers waiting on hold are position 1 in their respective queues. A queue show might look like this... A has 1 calls (max unlimited) in 'leastrecent' strategy (0s holdtime, 533s talktime), W:1, C:1, A:0, SL:100.0% within 60s Members: 21 (Local/21@from-queue/n) (dynamic) (In use) has taken 1 calls (last was 533 secs ago) Callers: 1. SIP/Trunk-eb17 (wait: 1:14, prio: 0) B has 1 calls (max unlimited) in 'leastrecent' strategy (0s holdtime, 533s talktime), W:1, C:1, A:0, SL:100.0% within 60s Members: 21 (Local/21@from-queue/n) (dynamic) (In use) has taken 1 calls (last was 533 secs ago) Callers: 1. SIP/Trunk-eb1e (wait: 0:45, prio: 0) My question is: when Alice gets off the phone, which call will she get? My expectation is that she will get the call which has been waiting longer, but I'm not sure that's actually the case. Alex Forster Depends on many factors, but if the queues have equal priority, the longest wait will (AFAIK) take priority. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 11.0.0-rc2 Now Available
The Asterisk Development Team has announced the second release candidate of Asterisk 11.0.0. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 11.0.0-rc2 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release candidate: * --- Fix an issue where outgoing calls would fail to establish audio due to ICE negotiation failures. (Closes issue ASTERISK-20554. Reported by mmichelson) * --- Ensure Asterisk fails TCP/TLS SIP calls when certificate checking fails (Closes issue ASTERISK-20559. Reported by kmoore) * --- Don't make chan_sip export global symbols. (Closes issue ASTERISK-20545. Reported by kmoore) For a full list of changes in this release candidate, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.0.0-rc2 Thank you for your continued support of Asterisk! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RC2, was: Motif/XMPP for Google Voice
Hi, With regards to: On Mon, 2012-10-15 at 09:09 -0500, Joshua Colp wrote: asterisk asterisk wrote: Dear all, Hola, I wish to ask a question of the new Motif Channel in asterisk 11. I successfully compile the binary and run without error. However, when dialing out, no external connection only ringing. During testing some issues were uncovered with the Motif channel driver, but unfortunately they did not make the last release candidate. My suggestion is to get Asterisk 11 from SVN or if you are not comfortable with that wait until the official Asterisk 11 release. Cheers, And to: Asterisk 11.0.0-rc2 Now Available skimming through http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.0.0-rc2 I did not see any reference towards Motif/XMPP. So your code is still only in SVN, not in the RC2? Hans -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] RC2, was: Motif/XMPP for Google Voice
Hans Witvliet wrote: And to: Asterisk 11.0.0-rc2 Now Available skimming through http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.0.0-rc2 I did not see any reference towards Motif/XMPP. So your code is still only in SVN, not in the RC2? The commits in question: * [r374850] Fix an issue where outgoing calls would fail to establish audio due to ICE negotiation failures. This change removes the requirement for ufrag and pwd in the transport stanza and also makes us the controlling agent. * [r374877] Fix a bug where audio on Google Voice would not work due to ignoring candidates. Instead of ignoring parts of the message that are not known just ignore the ones we know may be present and that would cause a problem. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Use of Sangoma D500
Does anyone on the list have any experience with using a Sangoma D500 card with Asterisk to transcode G729? If you could mention pros and cons I would like to hear opinions. Thanks -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Unable to load users.conf
Dear All, I've got this Warning message on my log: WARNING[3741]: res_phoneprov.c:923 set_config: Unable to load users.conf what is this mean? thank you. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unable to load users.conf
On Thu, 18 Oct 2012, Rizha Yuherdianto wrote: I've got this Warning message on my log: WARNING[3741]: res_phoneprov.c:923 set_config: Unable to load users.conf what is this mean? thank you. I'm just a 1.2 Luddite, but I'd guess: 0) It's just a warning so it may not be a big deal. 1) The file does not exist. 2) The file is not in the correct directory. 3) You have a 'permissions' issue. 4) The file is invalid. Maybe somebody clobbered the file by editing it with Notepad. -- 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 -- New to Asterisk? Join us for a live introductory webinar every 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] Unable to load users.conf
Thank you Steve, I'm using AsteriskNow latest version 2.0.2. my answer is: 0) if its just a warning, how to get it fixed? 1) checked, it is not exist. is it exist by default? 2) what directory it should be? 3) im root 4) the file doesn't exist Thanks On Thu, Oct 18, 2012 at 11:18 AM, Steve Edwards asterisk@sedwards.comwrote: On Thu, 18 Oct 2012, Rizha Yuherdianto wrote: I've got this Warning message on my log: WARNING[3741]: res_phoneprov.c:923 set_config: Unable to load users.conf what is this mean? thank you. I'm just a 1.2 Luddite, but I'd guess: 0) It's just a warning so it may not be a big deal. 1) The file does not exist. 2) The file is not in the correct directory. 3) You have a 'permissions' issue. 4) The file is invalid. Maybe somebody clobbered the file by editing it with Notepad. -- 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://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] Unable to load users.conf
On Thu, 18 Oct 2012, Rizha Yuherdianto wrote: 0) if its just a warning, how to get it fixed? It doesn't really need to. A 'warning' is like saying here's something you should be aware of. Personally, I prefer to resolve all warnings so there is less cruft to sift through when something actually does go wrong. 1) checked, it is not exist. is it exist by default? I don't know about your version of Asterisk. 2) what directory it should be? Unless you (or your package maintainer) has been fiddling about, it should be in the same directory as all of your other Asterisk configuration files: sip.conf, iax.conf, extensions.[conf|ael], etc. 3) im root Glad to meet you. If you meant the user running Asterisk is root, this is a less than optimal situation that can lead to really big problems. 4) the file doesn't exist At a minimum, 'touch /etc/asterisk/users.conf' may make the warning go away. You should read up a bit to see if the features of users.conf make sense for your environment. Personally, I set up my Asterisk installs so they only load the modules I'm actually using by specifying 'autoload=no' and explicitly loading the modules I want in modules.conf. -- 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 -- New to Asterisk? Join us for a live introductory webinar every 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] Unable to load users.conf
0) if its just a warning, how to get it fixed? It doesn't really need to. A 'warning' is like saying here's something you should be aware of. Personally, I prefer to resolve all warnings so there is less cruft to sift through when something actually does go wrong. I see 1) checked, it is not exist. is it exist by default? I don't know about your version of Asterisk. Im using AsteriskNow lastest version 2.0.2 2) what directory it should be? Unless you (or your package maintainer) has been fiddling about, it should be in the same directory as all of your other Asterisk configuration files: sip.conf, iax.conf, extensions.[conf|ael], etc. already checked. files for asterisk are on /etc/asterisk directory but theres no users.conf 3) im root Glad to meet you. :D If you meant the user running Asterisk is root, this is a less than optimal situation that can lead to really big problems. Why? Steve please explain. 4) the file doesn't exist At a minimum, 'touch /etc/asterisk/users.conf' may make the warning go away. You should read up a bit to see if the features of users.conf make sense for your environment. I'll read it again Personally, I set up my Asterisk installs so they only load the modules I'm actually using by specifying 'autoload=no' and explicitly loading the modules I want in modules.conf. I still need to learn more. I've just experimenting asterisk from yesterday, but I'll try to examine what modules i need to load :-) thank you -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Critical Asterisk Outage - Installing g729 crashed asterisk.
On Thu, Oct 18, 2012 at 1:35 AM, Jared Baxley jared.bax...@gmail.comwrote: I was following Digium's instructions to the letter to install g729. but upon telling asterisk to load the module, the system hung. after a few minutes later a CTRL-C and attempted to run the command again. Same result. any g729 show command returns nothing... no error no results. Reboot the server and asterisk will not process calls. Freepbx shows the following. [2012-Oct-18 01:21:04] [INFO] (bin/retrieve_conf:109) - found language dir fr for directory, not installed on system, skipping [2012-Oct-18 01:21:07] [FATAL] (libraries/utility.functions.php:429) - retreive_conf failed to get engine information and cannot configure up a softwitch with out it. Error: ERROR-UNABLE-TO-PARSE [2012-Oct-18 01:21:07] [CRITICAL] (admin/functions.inc.php:366) - [NOTIFICATION]-[freepbx]-[RCONFFAIL] - retrieve_conf failed, config not applied This seems to indicate that the g729 module is working [2012-10-18 01:21:04] NOTICE[4717] codec_g729a.c: G.729A transcoding module version 1.8.0_3.1.5, Copyright (C) 1999-2009 Digium, Inc. [2012-10-18 01:21:04] NOTICE[4717] codec_g729a.c: This module is supplied under a commercial license granted by Digium, Inc. [2012-10-18 01:21:04] NOTICE[4717] codec_g729a.c: Please see the full license text supplied by the accompanying [2012-10-18 01:21:04] NOTICE[4717] codec_g729a.c: register utility, or ask for a copy from Digium. [2012-10-18 01:21:04] NOTICE[4717] codec_g729a.c: This product includes software developed by the OpenSSL Project [2012-10-18 01:21:04] NOTICE[4717] codec_g729a.c: for use in the OpenSSL Toolkit. (http://www.openssl.org/) [2012-10-18 01:21:04] NOTICE[4717] codec_g729a.c: Copyright (C) 1998-2006 The OpenSSL Project [2012-10-18 01:21:04] VERBOSE[4717] manager.c: == Manager registered action G729LicenseStatus [2012-10-18 01:21:04] VERBOSE[4717] manager.c: == Manager registered action G729LicenseList [2012-10-18 01:21:04] VERBOSE[4717] codec_g729a.c: == Host-ID: 60:b6:d3:a0:c9:be:6e:46:74:41:07:b3:8e:76:59:b1:ba:5c:59:05 [2012-10-18 01:21:04] VERBOSE[4717] codec_g729a.c: == Found license 'G729-XXX' providing 40 channels [2012-10-18 01:21:04] VERBOSE[4717] codec_g729a.c: == Found total of 40 G.729 licenses How do i roll this back? Just delete codec_g729a.so ? You can do a noload in modules.conf. This doesn't appear to be the problem though. It may be. Did you try saving a change in FreePBX and applying it? It seems more like a FreePBX config error that should be overwritten by FreePBX database to flat files. Thanks, Steve Totaro -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] Critical Asterisk Outage - Installing g729 crashed asterisk.
On Thu, Oct 18, 2012 at 1:49 AM, Steve Totaro stot...@totarotechnologies.com wrote: On Thu, Oct 18, 2012 at 1:35 AM, Jared Baxley jared.bax...@gmail.comwrote: I was following Digium's instructions to the letter to install g729. but upon telling asterisk to load the module, the system hung. after a few minutes later a CTRL-C and attempted to run the command again. Same result. any g729 show command returns nothing... no error no results. Reboot the server and asterisk will not process calls. Freepbx shows the following. [2012-Oct-18 01:21:04] [INFO] (bin/retrieve_conf:109) - found language dir fr for directory, not installed on system, skipping [2012-Oct-18 01:21:07] [FATAL] (libraries/utility.functions.php:429) - retreive_conf failed to get engine information and cannot configure up a softwitch with out it. Error: ERROR-UNABLE-TO-PARSE [2012-Oct-18 01:21:07] [CRITICAL] (admin/functions.inc.php:366) - [NOTIFICATION]-[freepbx]-[RCONFFAIL] - retrieve_conf failed, config not applied This seems to indicate that the g729 module is working [2012-10-18 01:21:04] NOTICE[4717] codec_g729a.c: G.729A transcoding module version 1.8.0_3.1.5, Copyright (C) 1999-2009 Digium, Inc. [2012-10-18 01:21:04] NOTICE[4717] codec_g729a.c: This module is supplied under a commercial license granted by Digium, Inc. [2012-10-18 01:21:04] NOTICE[4717] codec_g729a.c: Please see the full license text supplied by the accompanying [2012-10-18 01:21:04] NOTICE[4717] codec_g729a.c: register utility, or ask for a copy from Digium. [2012-10-18 01:21:04] NOTICE[4717] codec_g729a.c: This product includes software developed by the OpenSSL Project [2012-10-18 01:21:04] NOTICE[4717] codec_g729a.c: for use in the OpenSSL Toolkit. (http://www.openssl.org/) [2012-10-18 01:21:04] NOTICE[4717] codec_g729a.c: Copyright (C) 1998-2006 The OpenSSL Project [2012-10-18 01:21:04] VERBOSE[4717] manager.c: == Manager registered action G729LicenseStatus [2012-10-18 01:21:04] VERBOSE[4717] manager.c: == Manager registered action G729LicenseList [2012-10-18 01:21:04] VERBOSE[4717] codec_g729a.c: == Host-ID: 60:b6:d3:a0:c9:be:6e:46:74:41:07:b3:8e:76:59:b1:ba:5c:59:05 [2012-10-18 01:21:04] VERBOSE[4717] codec_g729a.c: == Found license 'G729-XXX' providing 40 channels [2012-10-18 01:21:04] VERBOSE[4717] codec_g729a.c: == Found total of 40 G.729 licenses How do i roll this back? Just delete codec_g729a.so ? You can do a noload in modules.conf. This doesn't appear to be the problem though. It may be. Did you try saving a change in FreePBX and applying it? It seems more like a FreePBX config error that should be overwritten by FreePBX database to flat files. Thanks, Steve Totaro See here http://www.freepbx.org/forum/freepbx/users/apply-configuration-changes-errors-with-failed-to-get-engine-info-retreive-conf Very similar problem with FreePBX, your G729 looks fine. Check permissions and ownership of any files you changed. Thanks, Steve Totaro -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
