Re: [asterisk-users] Dialplan - working out when users answer
On Wed, Dec 19, 2012 at 12:44 PM, Andrew White
and...@computersforall.com.au wrote:
Hi Satish/list,
** **
Looks like I spoke to soon.
** **
I have the following in my dialplan:
** **
*Dial(${QUEUEEXTS},${RINGTIME},U(queueControl^direct^CONNECTED))*
** **
And after confirming with a “dialplan show” it was definitely in there, I
continued to get this:
** **
*ERROR[28167]: app_stack.c:420 gosub_exec: Attempt to reach a
non-existent destination for gosub: (Context:queueControl, Extension:s,
Priority:1)*
* *
I can’t quite work out why it would be trying to s/1 instead of
direct/CONNECTED =/.
** **
Any ideas?
** **
Thanks!
In your case, direct and CONNECTED have to be arguments and not the
extension and priority value respectively. Calling Subroutine from dial
will always start execution with extension s and priority 1.
See the link for more information, Arguments are passed to subroutine using
^ as a delimiter.
--Satish Barot
** **
*From:* Andrew White
*Sent:* Wednesday, 19 December 2012 5:58 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* RE: [asterisk-users] Dialplan - working out when users answer**
**
** **
Thanks Satish, fantastic advice. I didn’t even think to look into the dial
options – doh!
** **
Thanks very much,
** **
Andrew
** **
*From:* asterisk-users-boun...@lists.digium.com [
mailto:asterisk-users-boun...@lists.digium.comasterisk-users-boun...@lists.digium.com]
*On Behalf Of *Satish Barot
*Sent:* Wednesday, 19 December 2012 4:40 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Dialplan - working out when users answer**
**
** **
On Wed, Dec 19, 2012 at 10:53 AM, Andrew White
and...@computersforall.com.au wrote:
Hey guys,
I’ve got a part of my dialplan that dials multiple people:
*exten = direct,n,Dial(${QUEUEEXTS},${RINGTIME})
*Multiple extensions are in the ${QUEUEEXTS} from an external script –
e.g. SIP/100SIP/101SIP/105 etc
This works great, however I want to see if I can find a way to work out
(and run an AGI script) when the call is picked up by someone.
Thanks all!
Option M or U of Dial application would help you do this.
https://wiki.asterisk.org/wiki/display/AST/Application_Dial.
** **
--Satish Barot
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[asterisk-users] Unable to create channel of type 'SIP' (cause 20 - Unknown)
2012/12/19 Scott Huang gyration.hu...@gmail.com Hi I've saw some similar case in the mail list, but seems no standard answers, so I decide ask here again. Is there anyone see the message below ? I use asterisk(1.8.11-cert 9) in my openbts2.8, and when I made a phone call, the Asterisk CLI poppd the following messages. = *CLI == Using SIP RTP CoS mark 5 -- Executing [8690@phones:1] Dial(SIP/IMSI466974600011287-, SIP/IMSI466974104638690) in new stack [Dec 18 16:00:49] WARNING[2934]: app_dial.c:2218 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) -- Auto fallthrough, channel 'SIP/IMSI466974600011287-' status is 'CHANUNAVAIL' == The attached files are the sip.conf and extension.conf and wireshark trace log. The part of my setting in sip.conf is: [IMSI466974104638690]; callerid=8690 8690 ; regexten=8690; canreinvite=no type=friend allow=gsm context=phones host=dynamic registertrying=yes [IMSI466974102820333]; callerid=0333 0333 ; regexten=0333; canreinvite=no type=friend allow=gsm context=phones host=dynamic registertrying=yes [IMSI466974600011287]; callerid=1287 1287 ; regexten=1287; canreinvite=no type=friend allow=gsm context=phones host=dynamic registertrying=yes The part of my setting in extensions.conf is: [phones] exten = 8690,1,Dial(SIP/IMSI466974104638690) exten = 0333,1,Dial(SIP/IMSI466974102820333) exten = 1287,1,Dial(SIP/IMSI466974600011287) How to exactly configure asterisk for a sip call ? Thanks very much ! BR/Scott -- _ -- 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] Possible bug - queue doesn't play hold music
On Fri, 2012-12-14 at 15:16 +, Ishfaq Malik wrote: Hi Can someone else please check the following: We have installed asterisk 1.8.18.0 onto our development and test servers. They were previously on 1.8.7.0 When an inbound call executes a queue, I can see in the logs that the hold music is supposed to start playing but there is no sound. If the call is answered and the callee puts the caller on hold, I can see the same log message of hold music starting but this time the hold music can be heard. This is happening on both installations of 1.8.18.0. If other people are experiencing the same thing we can raise a bug on it. Log excerpts below with my comments after a # symbol -- Executing [s@ethn-xx-work:4] Queue(SIP/x.x.x.x-0061, test-ish,Tn,,,600) -- Started music on hold, class 'default', on SIP/x.x.x.x-0061 #comment: no music heard == Using SIP RTP CoS mark 5 -- SIP/101-0062 is ringing -- SIP/101-0062 is ringing -- SIP/101-0062 is ringing -- SIP/101-0062 is ringing -- SIP/101-0062 is ringing -- SIP/101-0062 answered SIP/x.x.x.x-0061 -- Stopped music on hold on SIP/x.x.x.x-0061 [2012-12-14 14:44:04] ERROR[26568]: chan_sip.c:29941 setup_srtp: No SRTP module loaded, can't setup SRTP session. -- Started music on hold, class 'default', on SIP/x.x.x.x-0061 #comment: music can be heard [2012-12-14 14:44:12] ERROR[26568]: chan_sip.c:29941 setup_srtp: No SRTP module loaded, can't setup SRTP session. -- Stopped music on hold on SIP/x.x.x.x-0061 == Spawn extension (ethn-xx-work, s, 4) exited non-zero on 'SIP/x.x.x.x-0061' Really could do with a second opinion on this issue as it would quite a serious bug if it is one... -- Ishfaq Malik i...@pack-net.co.uk Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, 2A ENTERPRISE HOUSE, LLOYD STREET NORTH, MANCHESTER SCIENCE PARK, MANCHESTER, M156SE COMPANY REG NO. 04920552 -- _ -- 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] Doubt regarding jabber
On Tue, 2012-12-18 at 10:01 -0600, Jonathan Rose wrote: Harish Mandowara wrote: I have Asterisk server 1.8.19 with jabber enabled. On the other side i have openfire server with asterisk-im enabled. I have a doubt, whether my sip client connected with asterisk can send message to other sip client, which is connected to same asterisk server. I have jitsi as a sip client. If its possible. Than please suggest any documentation regarding this. any help?? THanks a lot As far as I'm aware, SIP clients are generally incapable of using XMPP to send and receive messages. I'm aware Jitsi can act as an XMPP client, but its functionality as one has basically nothing to do with Asterisk. Asterisk can use XMPP send and receive messages to/from an XMPP server (also clients on that server by relay). Jabber is also used for Google Talk and Google Voice, but I'm not sure which versions those features work best in. I'd imagine 11 would be your best bet if you wanted that functionality since it has a bunch of Jabber improvements as well as chan_motif. So if you want Asterisk to send jitsi an IM, you need to set up account on an XMPP server for them to use (as well as profiles to connect with). Once you are sure you have Asterisk and the Jitsi client connected to the XMPP server, you can send the message with the dialplan application 'JabberSend' which takes arguments of account (which is the account you are using to send), jid (who is receiving the message) and the message itself. You can similarly receive messages on Asterisk by using the JABBER_RECEIVE function with similar arguments except no message and an optional timeout. Hi, probably Jonathan is correct. firstly, much changes/improvements has been made since 1.8.19 so you might give asterisk-11 a try, in which xmpp has got a lot of attention. Secondly, as you use Jitsi, why don't you use jitsi's xmpp capabilities? afaicr, the SIP-part of jitsi is only capable of simple, not xmpp. 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
[asterisk-users] asterisk conferencing |MEETME or app_conference
Hi All, I am googling from few days back for a conference utility which fulfill my below scenario ,Please give your suggestion to fulfill my need. I have a scenario where leader is giving a lecture and other participants are on mute mode... At the end of conference, when QA session begins, is there a way for participants to raise hands, if they have any questions so Leader can unmute them? Is this feature already there in Meetme conference, if there then how can I implement this? Is there another utility which works in above scenario,what about app_conference? Please suggest ... Thanks Regards, Pankaj Pandey +91-9990212758-- _ -- 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 create channel of type 'SIP' (cause 20 - Unknown)
Scott Huang wrote: Hi I've saw some similar case in the mail list, but seems no standard answers, so I decide ask here again. Is there anyone see the message below ? I use asterisk(1.8.11-cert 9) in my openbts2.8, and when I made a phone call, the Asterisk CLI poppd the following messages. = *CLI == Using SIP RTP CoS mark 5 -- Executing [8690@phones:1] Dial(SIP/IMSI466974600011287-, SIP/IMSI466974104638690) in new stack [Dec 18 16:00:49] WARNING[2934]: app_dial.c:2218 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) -- Auto fallthrough, channel 'SIP/IMSI466974600011287-' status is 'CHANUNAVAIL' == When you use a dynamic host type, the device needs to register to Asterisk in order to be dialed. Otherwise there is no way to for Asterisk to know what address to send the invite to and Asterisk will make chan_sip issue the cause 20 error you are seeing. If the device has a static IP and you don't want to deal with registration, you could always change the host to that IP address. Alternatively you could just figure out how to get your devices to register to your Asterisk server. -- Jonathan R. Rose Digium, Inc. | Software Engineer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct +1 256 428 6139 Check us out at: http://digium.com http://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] loop start vs. kewl start for T1 interface
Is there a doc somewhere that explains when to use loop start vs. kewl start in certain situations? I've seen in a lot of posts that most people just say 'use kewl start', but nobody really explains why. It seems that if kewl start should be used 100% of the time then why is there even an option for loop start? After some research, it seems like there are 4 main loop start setup options: 1. loop start (classic) 2. loop start with battery reversal 3. loop start with battery drop (kewl start) 4. loop start with tone detection Am I correct in assuming that when using kewl start that both these things happen? 1. If the far end disconnects, the code will look for battery drop 2. If the near end disconnects, the code will initiate a battery drop My question is, if I'm using a T1 channel bank, do these values matter on the T1 interface, or only on the channel bank itself. In other words, do the disconnect supervision signals cross the channel bank and end up in the T1 signal, or does the T1 signal use a different mechanism to signal a disconnect? Specifically, I'm trying to use an Adtran Total Access 850 with FXO cards, and fairly often I will get phantom calls. I believe this to be a result of a hangup not being detected correctly. On a side note, Is there some kind of 'learning' tool that I can run that monitors the line and displays these kinds of events during the duration of a call? For instance I could run the tool and then it could guide me through a series of 'I call you, you call me' scenarios and then the tool would tell me what settings to use? Justin Killen -- _ -- 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] Possible bug - queue doesn't play hold music
On Fri, 2012-12-14 at 15:16 +, Ishfaq Malik wrote: Hi Can someone else please check the following: We have installed asterisk 1.8.18.0 onto our development and test servers. They were previously on 1.8.7.0 When an inbound call executes a queue, I can see in the logs that the hold music is supposed to start playing but there is no sound. If the call is answered and the callee puts the caller on hold, I can see the same log message of hold music starting but this time the hold music can be heard. This is happening on both installations of 1.8.18.0. If other people are experiencing the same thing we can raise a bug on it. Log excerpts below with my comments after a # symbol -- Executing [s@ethn-xx-work:4] Queue(SIP/x.x.x.x-0061, test-ish,Tn,,,600) -- Started music on hold, class 'default', on SIP/x.x.x.x-0061 #comment: no music heard == Using SIP RTP CoS mark 5 -- SIP/101-0062 is ringing -- SIP/101-0062 is ringing -- SIP/101-0062 is ringing -- SIP/101-0062 is ringing -- SIP/101-0062 is ringing -- SIP/101-0062 answered SIP/x.x.x.x-0061 -- Stopped music on hold on SIP/x.x.x.x-0061 [2012-12-14 14:44:04] ERROR[26568]: chan_sip.c:29941 setup_srtp: No SRTP module loaded, can't setup SRTP session. -- Started music on hold, class 'default', on SIP/x.x.x.x-0061 #comment: music can be heard [2012-12-14 14:44:12] ERROR[26568]: chan_sip.c:29941 setup_srtp: No SRTP module loaded, can't setup SRTP session. -- Stopped music on hold on SIP/x.x.x.x-0061 == Spawn extension (ethn-xx-work, s, 4) exited non-zero on 'SIP/x.x.x.x-0061' Really could do with a second opinion on this issue as it would quite a serious bug if it is one... The incoming call leg does not appear to be answered yet so I would not expect the caller to be able to hear MOH. Richard -- _ -- 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] Catching hold in dialplan
I’ve built a custom application for our call center and am having one problem. Unfortunately certain things happen whilst the agent has the customer on hold which I’d like to work around. But I can’t work out how to catch the actual hold event so I can do something about it. From the console with verbosity on 12, all I can see is: -- Started music on hold, class 'default', on SIP/trunk-9546 -- Started music on hold, class 'default', on SIP/100-9547 I’m happy to try and catch this AGI or via manager if needed, however a dialplan based solution would be best. You will need to monitor AMI for MusicOnHold start/stop events. Dialplan normally executes before a call is connected/bridged. AGI is really an external form of dialplan execution. Richard -- _ -- 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] queues show some agents (In use) from the start
Hi gurus Some of my agents are in use with no call involved. Members: Local/0971230031@internal/n (In use) has taken no calls yet Local/0972105500@internal/n (In use) has taken no calls yet No Callers Is there a workarround so solve this? Wham am i doing wrong? If i change agent Local/0971230031 for an other mobile line works fine :( This is my config Asterisk 1.6.1.6 DAHDI Version: 2.6.1 libss7 version: 1.0.2 queues.conf [RCEN] musicclass = default strategy = rrmemory weight=0 wrapuptime=15 autopause=no setinterfacevar=yes setqueueentryvar=yes setqueuevar=yes eventwhencalled = yes ringinuse = no joinempty=yes member = Local/0972105500@internal/n member = Local/0971230031@internal/n Thanks in advance. rv -- _ -- 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] Congestion() forcing PRI channels to be not available
Hi, I have a PSTN Asterisk box that's connected to other dialplan PBXes through IAX2. Recently this box was upgraded to 1.4.44 with the latest DAHDI version. I've noticed that if one of the dialplan PBXes calls Congestion(), the PRI will return ISDN code 34 (as its supposed to do). However, the issue is that subsequent calls into that PRI channel are immediately responded by a Code 44 (channel not available) even though there is no live call on the channel. Has anyone else experienced this behavior? Its a pretty crippling behavior since all of our channels eventually become unresponsive until a 'dahdi restart' is issued. Thanks. -- James -- _ -- 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] Congestion() forcing PRI channels to be not available
You probably already know this, but 1.4x is very old (released in 2006) and is officially end-of-life. https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions You might get more help or better behavior by updating to a newer more current version of Asterisk, such as 1.8 which will be receiving bug fixes into October 2014. On Wed, Dec 19, 2012 at 3:47 PM, James Lamanna jlama...@gmail.com wrote: Hi, I have a PSTN Asterisk box that's connected to other dialplan PBXes through IAX2. Recently this box was upgraded to 1.4.44 with the latest DAHDI version. I've noticed that if one of the dialplan PBXes calls Congestion(), the PRI will return ISDN code 34 (as its supposed to do). However, the issue is that subsequent calls into that PRI channel are immediately responded by a Code 44 (channel not available) even though there is no live call on the channel. Has anyone else experienced this behavior? Its a pretty crippling behavior since all of our channels eventually become unresponsive until a 'dahdi restart' is issued. Thanks. -- James -- _ -- 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 -- -Chris Harrington ACSDi Office: 763.559.5800 Mobile Phone: 612.326.4248 -- _ -- 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 and DAHDI/i4
In 1.4.43 I would see things from core show channels like DAHDI/18/x for line 18 in Asterisk 11 its DAHDI/i4/ How do I get the line number back? 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
Re: [asterisk-users] asterisk 11 and DAHDI/i4
In 1.4.43 I would see things from core show channels like DAHDI/18/x for line 18 in Asterisk 11 its DAHDI/i4/ How do I get the line number back? This was a change in v1.8 and is documented in the v1.8 UPGRADE.txt file: * The PRI channels in chan_dahdi can no longer change the channel name if a different B channel is selected during call negotiation. To prevent using the channel name to infer what B channel a call is using and to avoid name collisions, the channel name format is changed. The new channel naming for PRI channels is: DAHDI/ispan/number[:subaddress]-sequence-number * Added CHANNEL(dahdi_span), CHANNEL(dahdi_channel), and CHANNEL(dahdi_type) so the dialplan can determine the B channel currently in use by the channel. Use CHANNEL(no_media_path) to determine if the channel even has a B channel. * Added AMI event DAHDIChannel to associate a DAHDI channel with an Asterisk channel so AMI applications can passively determine the B channel currently in use. Calls with no-media as the DAHDIChannel do not have an associated B channel. No-media calls are either on hold or call-waiting. Richard -- _ -- 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
