[asterisk-users] Asterisk Error
Hi all, Can you please let me know what the below issue mean when trying to start asterisk and how I can fix it? pbx_dundi.c: No ethernet interface found for seeding global EID You will have to set it manually. regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MoH - can the volume be adjusted
We have noticed MoH volume levels vary, very much depending on the terminal device that is connected. Within Asterisk is there any AGC or level control available to compensate for the varying terminal devices and their levels? For example a Polycom IP 7000 has very audible level while X Lite on a Dell laptop has very low level and Cisco 7970G have lower level than the Polycom IP 7000. What is the experience of other users and how have you handled this level variation? Rupert ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [HELP] - Conference bridge
Currently the conference bridge in Asterisk can be set for a maximum of 99 hours. For normal use this is more than adequate. However, we have a requirement to have the conference bridge permanently set up with no maximum time. Does anyone have experience on the possibility of changing this setting. Rupert ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Skill based routing
We are trying to implement skill based routing for agents in a support centre based on the agent login. Has anyone had any experience with this and what was the outcome? Can anyone share their ideas on this? Rupert ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [HELP] - Conference bridge
Rupert Utteridge - Digital Techniques (Austalia) Limited wrote: Currently the conference bridge in Asterisk can be set for a maximum of 99 hours. For normal use this is more than adequate. However, we have a requirement to have the conference bridge permanently set up with no maximum time. Where did you hear of this limitation? Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skill based routing
Rupert Utteridge - Digital Techniques (Austalia) Limited wrote: We are trying to implement skill based routing for agents in a support centre based on the agent login. Has anyone had any experience with this and what was the outcome? It can't really be done using Asterisk queues, unless you want to create a large number of queues for every relevant skill factor and have agents join various combinations of these simultaneously--which would take quite a bit of dial plan and/or AGI logic to pull off. Plus, that doesn't scale any given queue beyond one host. I suggest you look into using FastAGI[1] to simulate the queue experience by generating hold music and announcements without actually using Asterisk queues per se. This is quite possible to do, and, this allows you to distribute queues across multiple hosts, as well as distribute calls within those queues by whatever logic you choose. No shoehorning--just write it yourself. -- Alex [1] Yes, FastAGI. Not local AGI. And most especially not in PHP; contrary to a lot of the info out there, PHP could not possibly be a less suitable language in which to write AGI scripts. I don't know who comes up with these lavish heights of mediocrity. -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Error
This means that no ethernet interface is found for seeding the global EID. So you will have to set it manually. :) Pretty clear. On Thu, Jul 16, 2009 at 11:08 PM, michel freihamich...@gmail.com wrote: Hi all, Can you please let me know what the below issue mean when trying to start asterisk and how I can fix it? pbx_dundi.c: No ethernet interface found for seeding global EID You will have to set it manually. regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Error
I would guess that the MAC address of an Ethernet adaptor is used as a seed for a pseudorandom number generation algorithm that is used to create a GUID (Globally Unique Identifier) for your DUNDI node. But that requires an Ethernet adaptor. Ali Jawad wrote: This means that no ethernet interface is found for seeding the global EID. So you will have to set it manually. :) Pretty clear. On Thu, Jul 16, 2009 at 11:08 PM, michel freihamich...@gmail.com wrote: Hi all, Can you please let me know what the below issue mean when trying to start asterisk and how I can fix it? pbx_dundi.c: No ethernet interface found for seeding global EID You will have to set it manually. regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How do I create an IVR/Dial Group that works properly?
Hi all,
I am trying to understand how I can get a simple IVR scenario to work
properly (having already removed most of my hair...).
The basic requirement is as follows:
* Caller arrives at our main number
* Caller is greeted and then told they can enter an extension number, if
known, or wait and their call will be connected to an available rep.
* The IVR then dials a group of extensions (if the caller didn't enter
one obviously).
* Someone picks up the call and the connection is established and logged.
Now, I have all of this working apart from the last piece.
My IVR rings various extensions and I can pick up the call just fine.
But my problem is that the data asterisk records regarding the call is
wrong.
It correctly identifies the CallerID, but it always records the
destination as s. Not the extension of, for example my SIP phone (101).
If the incoming caller dials 101 whilst in the IVR, the log is correct.
I can see *why* I am having this problem (There is no extension when you
arrive in the IVR other than s), but I cannot see *how* to fix it.
Please can I ask how do others handle this so it works properly (I've
included the basics of my DP below)?
I'm running Asterisk 1.4.21.2~dfsg-1ubuntu3 on Ubuntu Server 8.10.
Thanks
Alan
Here is the IVR which callers are dropped into:
[tolc_menu] ; Welcome and information to callers
exten = s,1,Answer()
exten = s,n,Wait(2)
exten = s,n,Background(welcome-to-tolc) ; Say Hello
exten = s,n,Wait(1)
exten = s,n(tryagain),Background(enter-ext-of-personor) ; Enter
extension number if known, or
exten = s,n,Background(pls-stay-on-line) ; Trying to connect...
exten = s,n,WaitExten(5)
exten = s,n,Macro(belllord,${ALANL}${ALANB},303)
exten = _10[1-5],1,Macro(call_extension,SIP/${EXTEN})
exten = _20[1-5],1,Macro(call_extension,IAX2/alanb/${EXTEN})
The Vars ALANL and ALANB are:
ALANL=SIP/101
ALANB=IAX2/alanb/202
Here is the Macro belllord:
[macro-belllord]
exten = s,1,Dial(${ARG1},20,t)
exten = s,n,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Voicemail(${ar...@business,u) ; business is the
voicemail context, ${ARG2} is the mailbox number to dial
exten = s-NOANSWER,n,Hangup()
exten = s-BUSY,1,Voicemail(${ar...@business,b)
exten = s-BUSY,n,Hangup()
exten = _s-.,1,Goto(s-NOANSWER,1)
Here is the call-extension Macro:
[macro-call_extension]
exten = s,1,Dial(${ARG1},20,t) ; Ring channel for up to 20s
exten = s,n,Goto(s-${DIALSTATUS},1) ; Go to either no answer or busy.
exten = s-NOANSWER,1,Voicemail(${macro_ext...@garden_house,u)
exten = s-BUSY,1,Voicemail(${macro_ext...@garden_house,b)
exten = _s-.,1,Goto(s-NOANSWER,1)
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Re: [asterisk-users] Iphone setup
James Noble a écrit : Thank you for the heads up. I will look into both weephone and voipover3g I think siax -from cydia- could also be an alternative as they stated to use natively 3g. I only test WIFI. -- Daniel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] dialplan number matching
Hi, How can I match an extension ending with 3 (just an example but applicable to any other digit, including * or #)? exten = _ZX.3,n,... exten = _ZX.#,n,... (the above does not work) Can regular expressions be used in the standard dialplan (end with: $)? Thanks, Vieri ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is Enum safe from spammers?
Gordon Henderson schrieb: Just been contacted by a UK Enum registrar looking for ITSPs to become resellers of their Enum registration systems ... Is anyone using Enum? Yes. Does anyone (other than cynical old me) think that Enum is a spammers best friend? I think ENUM will not cause SPIT, but it can increase the efficiency. Has anyone received a spam VoIP call yet? (ie. one placed directly over the Internet aimed at a SIP URI to a PBX which allows anonymous incoming calls?) No. I can see that Enum is good to provide another way round the PSTN, but at the same time, I'm just not convinced... What do others think? SPIT (VoIP SPAM) is basically not a problem of ENUM, but of the communication protocol (SIP, H323, IAX, XMPP). E.g. SIP was developed with the same idea as SMTP: open connectivity - everybody can send a message to everyone with the need of peering agreements (thus, free of charge). Of course this introduces the same problems as SMTP has. Unfortunately the designers of SIP did not searched for a solution for this problem. Now, there is SIP-Identity which would allow (would, because nobody uses it) authentication of the caller - which is the basis for black/whitelists. H323 and IAX might be different, but they also allow to have unauthenticated calls. So, as soon as you operate your VoIP environment in a open way (regardless if it is SIP, XMPP ...) you are vulnerable to SPIT - even if you do not have ENUM provisioned for your local extensions. ENUM can be used by crawlers to find out valid VoIP URIs and can help SPITting, but in the end the problems is on the SIP level and must be solved there. regards klaus ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI hunt group
On Thursday, July 16, 2009, Alex Balashov wrote: C F wrote: If you don't want to port it to the PRI for whatever reason you can convert it to a RCFW (remote call forwarded number) which is around $15.00 plus $8.00 for each additional channel again pricing is for here in Verizon land. Is that true even if the number is out of a rate center that is billed long-distance relative to the destination (but still intra-LATA)? Or do you pay normal LD rates on top of all that in the intra-LATA LD scenario? Alex, Calls forwarded via Remote Call Forwarding are just like calls forwarded from a metered business or residential POTS line. If the destination to which you have selected to forward calls is normally a local call, you will just incur the standard metered call rate. If the call is normally a local toll charge (within the same LATA), you will incur toll charges from the LEC. If the call is long distance, you will need to select an IXC -- who will bill just as if the calls were made from a POTS line. Sincerely, Trevor Hammonds ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T38 negotiation, the last step !
Xavier Cardil schrieb: Hi, I've managed to get HYLAFAXT38MODEM- ASTERISKCISCOAS5400 working, but when they are negotiating asterisk drops a message telling Unknown RTP codec 96 received from gateway Do somebody know how to fix it ? Thank you ! [ TYPE: Control (4) SUBCLASS: Ringing (3) ] [SIP/GWCISCO5400O-600bfcc8] [ TYPE: Control (4) SUBCLASS: Answer (4) ] [SIP/GWCISCO5400O-600bfcc8] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/GWCISCO5400O-600bfcc8] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/GWCISCO5400O-600bfcc8] [ TYPE: Control (4) SUBCLASS: T38/Negotiation Requested (19) ] [SIP/GWCISCO5400O-600bfcc8] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/GWCISCO5400O-600bfcc8] [Jul 16 17:50:39] NOTICE[2736]: rtp.c:1739 ast_rtp_read: Unknown RTP codec 96 received from '192.168.3.163' Is '192.168.3.163' the Cisco GW? If yes, then it is probably an NSE packet. NSE is a Cisco proprietary FaxoverIP solution and uses per default payload types 96 and 97 to signal a changeover from VoIP to FoIP. Probably you have to configure the Cisco GW to use T.38 instead of NSE for FoIP. regards klaus ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI hunt group
Understood--thanks Trevor. I had wondered if the need to pay per channel might somehow amortize the LD balance. Appreciate your clarification. -- Sent from mobile device On Jul 17, 2009, at 5:14 AM, Trevor Hammonds tre...@concipient.net wrote: On Thursday, July 16, 2009, Alex Balashov wrote: C F wrote: If you don't want to port it to the PRI for whatever reason you can convert it to a RCFW (remote call forwarded number) which is around $15.00 plus $8.00 for each additional channel again pricing is for here in Verizon land. Is that true even if the number is out of a rate center that is billed long-distance relative to the destination (but still intra-LATA)? Or do you pay normal LD rates on top of all that in the intra-LATA LD scenario? Alex, Calls forwarded via Remote Call Forwarding are just like calls forwarded from a metered business or residential POTS line. If the destination to which you have selected to forward calls is normally a local call, you will just incur the standard metered call rate. If the call is normally a local toll charge (within the same LATA), you will incur toll charges from the LEC. If the call is long distance, you will need to select an IXC -- who will bill just as if the calls were made from a POTS line. Sincerely, Trevor Hammonds ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is Enum safe from spammers?
IMHO, anonymous calls should never, ever be accepted for a variety of reasons. It is naive. Just because it is convenient does not mean it should be done. Trusted calls between indeterminate parties can be arranged through peering federations, clearinghouses, etc. -- whatever VoIP peering model the market ultimately ends up adopting. -- Sent from mobile device On Jul 17, 2009, at 5:13 AM, Klaus Darilion klaus.mailingli...@pernau.at wrote: Gordon Henderson schrieb: Just been contacted by a UK Enum registrar looking for ITSPs to become resellers of their Enum registration systems ... Is anyone using Enum? Yes. Does anyone (other than cynical old me) think that Enum is a spammers best friend? I think ENUM will not cause SPIT, but it can increase the efficiency. Has anyone received a spam VoIP call yet? (ie. one placed directly over the Internet aimed at a SIP URI to a PBX which allows anonymous incoming calls?) No. I can see that Enum is good to provide another way round the PSTN, but at the same time, I'm just not convinced... What do others think? SPIT (VoIP SPAM) is basically not a problem of ENUM, but of the communication protocol (SIP, H323, IAX, XMPP). E.g. SIP was developed with the same idea as SMTP: open connectivity - everybody can send a message to everyone with the need of peering agreements (thus, free of charge). Of course this introduces the same problems as SMTP has. Unfortunately the designers of SIP did not searched for a solution for this problem. Now, there is SIP-Identity which would allow (would, because nobody uses it) authentication of the caller - which is the basis for black/whitelists. H323 and IAX might be different, but they also allow to have unauthenticated calls. So, as soon as you operate your VoIP environment in a open way (regardless if it is SIP, XMPP ...) you are vulnerable to SPIT - even if you do not have ENUM provisioned for your local extensions. ENUM can be used by crawlers to find out valid VoIP URIs and can help SPITting, but in the end the problems is on the SIP level and must be solved there. regards klaus ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and several clients behind NAT
Thanks Alex for your explanation. Does this NAT-mapping means that TAPI would also be possible ?? Jonas. On Tue, 2009-07-14 at 06:33 -0400, Alex Balashov wrote: Yes, this problem has a solution. The NAT gateway creates a UDP state mapping between internal source ports and external source (and destination, since most user agents are symmetrical nowadays) ports. The NAT gateway then allocates different external UDP ports for different connections being tracked in this manner. Consider, for example, two phones - 192.168.1.10 and 192.168.1.11 - registering to an outside SIP UAS through a NAT gateway whose public address is 67.194.23.55. The NAT gateway maps the source ports in a random or pseudorandom manner akin to: 192.168.1.10:5060 -- 67.194.23.55:32947 192.168.1.11:5060 -- 67.194.23.55:47948 If far-end NAT traversal is enabled on the UAS (in the case of Asterisk, that's nat=yes in sip.conf), the Contact URI supplied in the REGISTER message is ignored and the actual received IP and port on the network and transport layer is used in its place. The latter is what is stored as the contact binding. Later, a call comes in and the UAS maps it back to 67.194.23.55:47948 or 32947 depending on which registrant it is destined to go to. This scenario is not without its problems. Some user agents do not behave symmetrically. Some firewall/NAT router ALGs (application layer gateways) break this process, though they mean well and try to be helpful. But by far the most pressing problem is that many NAT gateways rather quickly age the temporary state information (internal:external UDP port mapping) out after a relatively short period of inactivity. That is why many far-end NAT traversal approaches implement a policy of periodically pinging the stored (received) contact with some sort of message that causes a bidirectional exchange of communication, and therefore causes the NAT gateway to reset its expiration timer for that connection state. In Asterisk, the OPTIONS messages generated when the qualify=yes option is enabled in sip.conf fulfill this function. Hope that helps, ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and several clients behind NAT
You're welcome. What's TAPI? -- Sent from mobile device On Jul 17, 2009, at 5:38 AM, jonas kellens jonas.kell...@telenet.be wrote: Thanks Alex for your explanation. Does this NAT-mapping means that TAPI would also be possible ?? Jonas. On Tue, 2009-07-14 at 06:33 -0400, Alex Balashov wrote: Yes, this problem has a solution. The NAT gateway creates a UDP state mapping between internal source ports and external source (and destination, since most user agents are symmetrical nowadays) ports. The NAT gateway then allocates different external UDP ports for different connections being tracked in this manner. Consider, for example, two phones - 192.168.1.10 and 192.168.1.11 - registering to an outside SIP UAS through a NAT gateway whose public address is 67.194.23.55. The NAT gateway maps the source ports in a random or pseudorandom manner akin to: 192.168.1.10:5060 -- 67.194.23.55:32947 192.168.1.11:5060 -- 67.194.23.55:47948 If far-end NAT traversal is enabled on the UAS (in the case of Asterisk, that's nat=yes in sip.conf), the Contact URI supplied in the REGISTER message is ignored and the actual received IP and port on the network and transport layer is used in its place. The latter is what is stored as the contact binding. Later, a call comes in and the UAS maps it back to 67.194.23.55:47948 or 32947 depending on which registrant it is destined to go to. This scenario is not without its problems. Some user agents do not behave symmetrically. Some firewall/NAT router ALGs (application layer gateways) break this process, though they mean well and try to be helpful. But by far the most pressing problem is that many NAT gateways rather quickly age the temporary state information (internal:external UDP port mapping) out after a relatively short period of inactivity. That is why many far-end NAT traversal approaches implement a policy of periodically pinging the stored (received) contact with some sort of message that causes a bidirectional exchange of communication, and therefore causes the NAT gateway to reset its expiration timer for that connection state. In Asterisk, the OPTIONS messages generated when the qualify=yes option is enabled in sip.conf fulfill this function. Hope that helps, ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skill based routing
On 7/17/09, Alex Balashov abalas...@evaristesys.com wrote: Rupert Utteridge - Digital Techniques (Austalia) Limited wrote: We are trying to implement skill based routing for agents in a support centre based on the agent login. Has anyone had any experience with this and what was the outcome? It can't really be done using Asterisk queues, unless you want to create a large number of queues for every relevant skill factor and have agents join various combinations of these simultaneously--which would take quite a bit of dial plan and/or AGI logic to pull off. Plus, that doesn't scale any given queue beyond one host. I suggest you look into using FastAGI[1] to simulate the queue experience by generating hold music and announcements without actually using Asterisk queues per se. This is quite possible to do, and, this allows you to distribute queues across multiple hosts, as well as distribute calls within those queues by whatever logic you choose. No shoehorning--just write it yourself. -- Alex [1] Yes, FastAGI. Not local AGI. And most especially not in PHP; contrary to a lot of the info out there, PHP could not possibly be a less suitable language in which to write AGI scripts. I don't know who comes up with these lavish heights of mediocrity. If you are not looking to write it yourself you could always try ViciDial which has skills-based routing built in, and it's free and Open Source. MATT--- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Queue member (Agent) does not Dial
Hi All, We are using Asterisk 1.2.18 in a CentOS box. Implemented a queue (maqueue) structure for handling customer calls. There are 4 queue members (85744,85766,85511,84888). These 4 members are logged in using AgentCallbackLogin application. But at some point, one of the agent's SIP phone does not ring for an incoming call to this queue. I checked the agent status and it is not in paused state. When I looked in the CLI, I couldn't see any attempt by the Asterisk to dial that particular agent. What are the possiblities for a queue member not dialed by Asterisk? This agent is defined in agents.conf, member of the queue defined in queues.conf and is not paused. The output of show agents from CLI is shown below: 8557 (Name1) available at '8...@specagentdial' (musiconhold is 'default') 8545 (Name2) not logged in (musiconhold is 'default') 8555 (Name3) available at '8...@specagentdial' (musiconhold is 'default') 8552 (Name4) not logged in (musiconhold is 'default') 8551 (Name5) not logged in (musiconhold is 'default') 8541 (Name6) not logged in (musiconhold is 'default') 8444 (Name7) not logged in (musiconhold is 'default') 85577(Name8) not logged in (musiconhold is 'default') 85744(Name9) available at '85...@specagentdial' (musiconhold is 'default') 85766(Name10) available at '85...@specagentdial' (musiconhold is 'default') 84888(Name11) available at '84...@specagentdial' (musiconhold is 'default') 85511(Name12) available at '85...@specagentdial' (musiconhold is 'default') The CLI message is given below: -- Executing Wait(Zap/1-1, 2) in new stack -- Executing Answer(Zap/1-1, ) in new stack -- Executing Playback(Zap/1-1, Thankyou9800) in new stack -- Executing Set(Zap/1-1, editeduid1=1247824046) in new stack -- Executing Set(Zap/1-1, editeduid2=897) in new stack -- Executing Set(Zap/1-1, editeduid=1247824046-897) in new stack -- Executing Set(Zap/1-1, MONITOR_FILENAME=QMA_20090717-054744_1247824046-897) in new stack -- Executing AGI(Zap/1-1, agi_queue.sh|QMA_20090717-054744_1247824046-897|MAQ) in new stack -- Executing Queue(Zap/1-1, maqueue|t|||180) in new stack -- Executing AGI(Local/84...@specagentdial-14bb,2, agi_qdial.sh|84888|315362) in new stack -- Executing AGI(Local/85...@specagentdial-beba,2, agi_qdial.sh|85744|315362) in new stack -- Executing AGI(Local/85...@specagentdial-67be,2, agi_qdial.sh|85511|315362) in new stack Here, from above, AGI program agi_qdial.sh which handles the dial operation does not make any attempt to dial 85766. Wondering why this is happening. The issue gets resolved only when asterisk service is restarted which is not a pretty good workaround. Any clue on this? Regards, Kurian Thayil. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skill based routing
Another simple way is to add local/foo/n as the only agent on the queue. In the dialplan for local/foo , interrogate a database for the most appropriate agent and then call that agent's extension. Julian 2009/7/17 Matt Florell astma...@gmail.com: On 7/17/09, Alex Balashov abalas...@evaristesys.com wrote: Rupert Utteridge - Digital Techniques (Austalia) Limited wrote: We are trying to implement skill based routing for agents in a support centre based on the agent login. Has anyone had any experience with this and what was the outcome? It can't really be done using Asterisk queues, unless you want to create a large number of queues for every relevant skill factor and have agents join various combinations of these simultaneously--which would take quite a bit of dial plan and/or AGI logic to pull off. Plus, that doesn't scale any given queue beyond one host. I suggest you look into using FastAGI[1] to simulate the queue experience by generating hold music and announcements without actually using Asterisk queues per se. This is quite possible to do, and, this allows you to distribute queues across multiple hosts, as well as distribute calls within those queues by whatever logic you choose. No shoehorning--just write it yourself. -- Alex [1] Yes, FastAGI. Not local AGI. And most especially not in PHP; contrary to a lot of the info out there, PHP could not possibly be a less suitable language in which to write AGI scripts. I don't know who comes up with these lavish heights of mediocrity. If you are not looking to write it yourself you could always try ViciDial which has skills-based routing built in, and it's free and Open Source. MATT--- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skill based routing
The simplicity of this approach is elegant, but in that case, why use a queue? Why not just perform this logic straight in the dial plan when processing the received call? The benefit of queues arises from their ability to keep state; they can retry agents, carry out different ring strategies, etc. I understood the original question to be implicitly about incorporating weights for skills into queue or queue-like call distribution mechanisms, since that is how it is done in call center products. If the question is simply how to make Asterisk consider certain outside information when choosing to whom to route a call, the answer would be that it is identical to the process for embedding any other kind of logic and/or outside data source into call processing. Another simple way is to add local/foo/n as the only agent on the queue. In the dialplan for local/foo , interrogate a database for the most appropriate agent and then call that agent's extension. Julian 2009/7/17 Matt Florell astma...@gmail.com: On 7/17/09, Alex Balashov abalas...@evaristesys.com wrote: Rupert Utteridge - Digital Techniques (Austalia) Limited wrote: We are trying to implement skill based routing for agents in a support centre based on the agent login. Has anyone had any experience with this and what was the outcome? It can't really be done using Asterisk queues, unless you want to create a large number of queues for every relevant skill factor and have agents join various combinations of these simultaneously--which would take quite a bit of dial plan and/or AGI logic to pull off. Plus, that doesn't scale any given queue beyond one host. I suggest you look into using FastAGI[1] to simulate the queue experience by generating hold music and announcements without actually using Asterisk queues per se. This is quite possible to do, and, this allows you to distribute queues across multiple hosts, as well as distribute calls within those queues by whatever logic you choose. No shoehorning--just write it yourself. -- Alex [1] Yes, FastAGI. Not local AGI. And most especially not in PHP; contrary to a lot of the info out there, PHP could not possibly be a less suitable language in which to write AGI scripts. I don't know who comes up with these lavish heights of mediocrity. If you are not looking to write it yourself you could always try ViciDial which has skills-based routing built in, and it's free and Open Source. MATT--- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skill based routing
We use a queue so that we can have all the benefits of the queue whilst finding an agent : music on hold, periodic announcements etc etc. You are right - with a little more effort we could probably remove the need for the queue. But why would I do that if I can use the queue for the bits I want ;) Julian 2009/7/17 Alex Balashov abalas...@evaristesys.com: The simplicity of this approach is elegant, but in that case, why use a queue? Why not just perform this logic straight in the dial plan when processing the received call? The benefit of queues arises from their ability to keep state; they can retry agents, carry out different ring strategies, etc. I understood the original question to be implicitly about incorporating weights for skills into queue or queue-like call distribution mechanisms, since that is how it is done in call center products. If the question is simply how to make Asterisk consider certain outside information when choosing to whom to route a call, the answer would be that it is identical to the process for embedding any other kind of logic and/or outside data source into call processing. Another simple way is to add local/foo/n as the only agent on the queue. In the dialplan for local/foo , interrogate a database for the most appropriate agent and then call that agent's extension. Julian 2009/7/17 Matt Florell astma...@gmail.com: On 7/17/09, Alex Balashov abalas...@evaristesys.com wrote: Rupert Utteridge - Digital Techniques (Austalia) Limited wrote: We are trying to implement skill based routing for agents in a support centre based on the agent login. Has anyone had any experience with this and what was the outcome? It can't really be done using Asterisk queues, unless you want to create a large number of queues for every relevant skill factor and have agents join various combinations of these simultaneously--which would take quite a bit of dial plan and/or AGI logic to pull off. Plus, that doesn't scale any given queue beyond one host. I suggest you look into using FastAGI[1] to simulate the queue experience by generating hold music and announcements without actually using Asterisk queues per se. This is quite possible to do, and, this allows you to distribute queues across multiple hosts, as well as distribute calls within those queues by whatever logic you choose. No shoehorning--just write it yourself. -- Alex [1] Yes, FastAGI. Not local AGI. And most especially not in PHP; contrary to a lot of the info out there, PHP could not possibly be a less suitable language in which to write AGI scripts. I don't know who comes up with these lavish heights of mediocrity. If you are not looking to write it yourself you could always try ViciDial which has skills-based routing built in, and it's free and Open Source. MATT--- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skill based routing
What value do the queue announcements (I am assuming these are pertaining to expected hold time, etc.) if there is only one agent? We use a queue so that we can have all the benefits of the queue whilst finding an agent : music on hold, periodic announcements etc etc. You are right - with a little more effort we could probably remove the need for the queue. But why would I do that if I can use the queue for the bits I want ;) Julian 2009/7/17 Alex Balashov abalas...@evaristesys.com: The simplicity of this approach is elegant, but in that case, why use a queue? Why not just perform this logic straight in the dial plan when processing the received call? The benefit of queues arises from their ability to keep state; they can retry agents, carry out different ring strategies, etc. I understood the original question to be implicitly about incorporating weights for skills into queue or queue-like call distribution mechanisms, since that is how it is done in call center products. If the question is simply how to make Asterisk consider certain outside information when choosing to whom to route a call, the answer would be that it is identical to the process for embedding any other kind of logic and/or outside data source into call processing. Another simple way is to add local/foo/n as the only agent on the queue. In the dialplan for local/foo , interrogate a database for the most appropriate agent and then call that agent's extension. Julian 2009/7/17 Matt Florell astma...@gmail.com: On 7/17/09, Alex Balashov abalas...@evaristesys.com wrote: Rupert Utteridge - Digital Techniques (Austalia) Limited wrote: We are trying to implement skill based routing for agents in a support centre based on the agent login. Has anyone had any experience with this and what was the outcome? It can't really be done using Asterisk queues, unless you want to create a large number of queues for every relevant skill factor and have agents join various combinations of these simultaneously--which would take quite a bit of dial plan and/or AGI logic to pull off. Plus, that doesn't scale any given queue beyond one host. I suggest you look into using FastAGI[1] to simulate the queue experience by generating hold music and announcements without actually using Asterisk queues per se. This is quite possible to do, and, this allows you to distribute queues across multiple hosts, as well as distribute calls within those queues by whatever logic you choose. No shoehorning--just write it yourself. -- Alex [1] Yes, FastAGI. Not local AGI. And most especially not in PHP; contrary to a lot of the info out there, PHP could not possibly be a less suitable language in which to write AGI scripts. I don't know who comes up with these lavish heights of mediocrity. If you are not looking to write it yourself you could always try ViciDial which has skills-based routing built in, and it's free and Open Source. MATT--- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skill based routing
Um, I really don't know - we just use the periodic messages to play the traditional Your call is important to use (whatever the wording..) Julian. 2009/7/17 Alex Balashov abalas...@evaristesys.com: What value do the queue announcements (I am assuming these are pertaining to expected hold time, etc.) if there is only one agent? We use a queue so that we can have all the benefits of the queue whilst finding an agent : music on hold, periodic announcements etc etc. You are right - with a little more effort we could probably remove the need for the queue. But why would I do that if I can use the queue for the bits I want ;) Julian 2009/7/17 Alex Balashov abalas...@evaristesys.com: The simplicity of this approach is elegant, but in that case, why use a queue? Why not just perform this logic straight in the dial plan when processing the received call? The benefit of queues arises from their ability to keep state; they can retry agents, carry out different ring strategies, etc. I understood the original question to be implicitly about incorporating weights for skills into queue or queue-like call distribution mechanisms, since that is how it is done in call center products. If the question is simply how to make Asterisk consider certain outside information when choosing to whom to route a call, the answer would be that it is identical to the process for embedding any other kind of logic and/or outside data source into call processing. Another simple way is to add local/foo/n as the only agent on the queue. In the dialplan for local/foo , interrogate a database for the most appropriate agent and then call that agent's extension. Julian 2009/7/17 Matt Florell astma...@gmail.com: On 7/17/09, Alex Balashov abalas...@evaristesys.com wrote: Rupert Utteridge - Digital Techniques (Austalia) Limited wrote: We are trying to implement skill based routing for agents in a support centre based on the agent login. Has anyone had any experience with this and what was the outcome? It can't really be done using Asterisk queues, unless you want to create a large number of queues for every relevant skill factor and have agents join various combinations of these simultaneously--which would take quite a bit of dial plan and/or AGI logic to pull off. Plus, that doesn't scale any given queue beyond one host. I suggest you look into using FastAGI[1] to simulate the queue experience by generating hold music and announcements without actually using Asterisk queues per se. This is quite possible to do, and, this allows you to distribute queues across multiple hosts, as well as distribute calls within those queues by whatever logic you choose. No shoehorning--just write it yourself. -- Alex [1] Yes, FastAGI. Not local AGI. And most especially not in PHP; contrary to a lot of the info out there, PHP could not possibly be a less suitable language in which to write AGI scripts. I don't know who comes up with these lavish heights of mediocrity. If you are not looking to write it yourself you could always try ViciDial which has skills-based routing built in, and it's free and Open Source. MATT--- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To
[asterisk-users] Friday reminder
Please join us today at 9AM PDT, 12 Noon EDT for the VoIP Users Conference to talk about the latest news and events in the wonderful world of VoIP. IRC #voip-users-conference SIP 7463#2262...@proxy.ideasip.com for g711 SIP 200...@login.zipdx.com (for g722 wideband-capable devices) See http://VUC.me for more details /r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Error
On Fri, Jul 17, 2009 at 2:08 AM, michel freiha mich...@gmail.com wrote: Hi all, Can you please let me know what the below issue mean when trying to start asterisk and how I can fix it? pbx_dundi.c: No ethernet interface found for seeding global EID You will have to set it manually. regards Add: noload = dundi To your modules.conf. That should fix it. Do you want to use dundi? What does ifconfig say? I assume you have a NIC? Is it up and all that when you start Asterisk? Have you tried downing it, setting all the variables (maybe even the MAC to be thorough) and then bringing it back up before starting Asterisk? Otherwise what kind of NIC? Do you have an old 3Com laying around you can pop in it? Open a bug report? -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Error
Dear Sir I did what you asked me to do...i added the following to /etc/opt/asterisk/modules.conf noload = dundi -bash-3.00# ifconfig -a lo0: flags=2001000849UP,LOOPBACK,RUNNING,MULTICAST,IPv4,VIRTUAL mtu 8232 index 1 inet 127.0.0.1 netmask ff00 eri0: flags=1000843UP,BROADCAST,RUNNING,MULTICAST,IPv4 mtu 1500 index 2 inet 192.168.0.178 netmask ff00 broadcast 192.168.0.255 ether 0:3:ba:f2:d2:ea Yes I have a NIC, Up and running and I can SSH the server from that NIC Regards On Fri, Jul 17, 2009 at 3:21 PM, Steve Totaro stot...@asteriskhelpdesk.comwrote: On Fri, Jul 17, 2009 at 2:08 AM, michel freiha mich...@gmail.com wrote: Hi all, Can you please let me know what the below issue mean when trying to start asterisk and how I can fix it? pbx_dundi.c: No ethernet interface found for seeding global EID You will have to set it manually. regards Add: noload = dundi To your modules.conf. That should fix it. Do you want to use dundi? What does ifconfig say? I assume you have a NIC? Is it up and all that when you start Asterisk? Have you tried downing it, setting all the variables (maybe even the MAC to be thorough) and then bringing it back up before starting Asterisk? Otherwise what kind of NIC? Do you have an old 3Com laying around you can pop in it? Open a bug report? -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dialplan number matching
Assuming you are using 4 digit extensions, this syntax would be: - exten = _ZXX3,n,... For 3 digits - exten = _ZX3,n,... The . is a wildcard that says take rest of number, so anything after that is irrelevant. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vieri Sent: Friday, July 17, 2009 4:11 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] dialplan number matching Hi, How can I match an extension ending with 3 (just an example but applicable to any other digit, including * or #)? exten = _ZX.3,n,... exten = _ZX.#,n,... (the above does not work) Can regular expressions be used in the standard dialplan (end with: $)? Thanks, Vieri ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skill based routing
Just use FastAGI to hit a little process that queries a database and returns the extensions of the most skilled Have FastAGI return those extensions and pass them to a dial command with the m flag (music if memory serves me correctly (pre-coffee) Make the music the standard junk you hear while waiting in a queue. Be sure the extensions you are calling don't have voicemail and you are obviously going to lose round robin and other dialing schemes unless you come up with some other logic. Multiple queues seems like the approach I would take. AMI can change agent penalties on the fly as well as add and remove them from queues. Just an FYI. Maybe a mix of multiple queues, FastAGI, dialplan logic, and AMI. Should be a fun project to get working and then fine tune. Thanks, Steve Totaro On Fri, Jul 17, 2009 at 6:36 AM, Julian Lyndon-Smith aster...@dotr.comwrote: Um, I really don't know - we just use the periodic messages to play the traditional Your call is important to use (whatever the wording..) Julian. 2009/7/17 Alex Balashov abalas...@evaristesys.com: What value do the queue announcements (I am assuming these are pertaining to expected hold time, etc.) if there is only one agent? We use a queue so that we can have all the benefits of the queue whilst finding an agent : music on hold, periodic announcements etc etc. You are right - with a little more effort we could probably remove the need for the queue. But why would I do that if I can use the queue for the bits I want ;) Julian 2009/7/17 Alex Balashov abalas...@evaristesys.com: The simplicity of this approach is elegant, but in that case, why use a queue? Why not just perform this logic straight in the dial plan when processing the received call? The benefit of queues arises from their ability to keep state; they can retry agents, carry out different ring strategies, etc. I understood the original question to be implicitly about incorporating weights for skills into queue or queue-like call distribution mechanisms, since that is how it is done in call center products. If the question is simply how to make Asterisk consider certain outside information when choosing to whom to route a call, the answer would be that it is identical to the process for embedding any other kind of logic and/or outside data source into call processing. Another simple way is to add local/foo/n as the only agent on the queue. In the dialplan for local/foo , interrogate a database for the most appropriate agent and then call that agent's extension. Julian 2009/7/17 Matt Florell astma...@gmail.com: On 7/17/09, Alex Balashov abalas...@evaristesys.com wrote: Rupert Utteridge - Digital Techniques (Austalia) Limited wrote: We are trying to implement skill based routing for agents in a support centre based on the agent login. Has anyone had any experience with this and what was the outcome? It can't really be done using Asterisk queues, unless you want to create a large number of queues for every relevant skill factor and have agents join various combinations of these simultaneously--which would take quite a bit of dial plan and/or AGI logic to pull off. Plus, that doesn't scale any given queue beyond one host. I suggest you look into using FastAGI[1] to simulate the queue experience by generating hold music and announcements without actually using Asterisk queues per se. This is quite possible to do, and, this allows you to distribute queues across multiple hosts, as well as distribute calls within those queues by whatever logic you choose. No shoehorning--just write it yourself. -- Alex [1] Yes, FastAGI. Not local AGI. And most especially not in PHP; contrary to a lot of the info out there, PHP could not possibly be a less suitable language in which to write AGI scripts. I don't know who comes up with these lavish heights of mediocrity. If you are not looking to write it yourself you could always try ViciDial which has skills-based routing built in, and it's free and Open Source. MATT--- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775 ___ -- Bandwidth and Colocation Provided
Re: [asterisk-users] How do I create an IVR/Dial Group that worksproperly?
I may 100% off here, but I seem to recall reading in the last 2 days threads
that macro dialing messes with CDR entries. I would try replacing one of
your macro lines with a straight Dial command to verify this.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alan Lord
(News)
Sent: Friday, July 17, 2009 3:23 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] How do I create an IVR/Dial Group that
worksproperly?
Hi all,
I am trying to understand how I can get a simple IVR scenario to work
properly (having already removed most of my hair...).
The basic requirement is as follows:
* Caller arrives at our main number
* Caller is greeted and then told they can enter an extension number, if
known, or wait and their call will be connected to an available rep.
* The IVR then dials a group of extensions (if the caller didn't enter
one obviously).
* Someone picks up the call and the connection is established and logged.
Now, I have all of this working apart from the last piece.
My IVR rings various extensions and I can pick up the call just fine.
But my problem is that the data asterisk records regarding the call is
wrong.
It correctly identifies the CallerID, but it always records the
destination as s. Not the extension of, for example my SIP phone (101).
If the incoming caller dials 101 whilst in the IVR, the log is correct.
I can see *why* I am having this problem (There is no extension when you
arrive in the IVR other than s), but I cannot see *how* to fix it.
Please can I ask how do others handle this so it works properly (I've
included the basics of my DP below)?
I'm running Asterisk 1.4.21.2~dfsg-1ubuntu3 on Ubuntu Server 8.10.
Thanks
Alan
Here is the IVR which callers are dropped into:
[tolc_menu] ; Welcome and information to callers
exten = s,1,Answer()
exten = s,n,Wait(2)
exten = s,n,Background(welcome-to-tolc) ; Say Hello
exten = s,n,Wait(1)
exten = s,n(tryagain),Background(enter-ext-of-personor) ; Enter
extension number if known, or
exten = s,n,Background(pls-stay-on-line) ; Trying to connect...
exten = s,n,WaitExten(5)
exten = s,n,Macro(belllord,${ALANL}${ALANB},303)
exten = _10[1-5],1,Macro(call_extension,SIP/${EXTEN})
exten = _20[1-5],1,Macro(call_extension,IAX2/alanb/${EXTEN})
The Vars ALANL and ALANB are:
ALANL=SIP/101
ALANB=IAX2/alanb/202
Here is the Macro belllord:
[macro-belllord]
exten = s,1,Dial(${ARG1},20,t)
exten = s,n,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Voicemail(${ar...@business,u) ; business is the
voicemail context, ${ARG2} is the mailbox number to dial
exten = s-NOANSWER,n,Hangup()
exten = s-BUSY,1,Voicemail(${ar...@business,b)
exten = s-BUSY,n,Hangup()
exten = _s-.,1,Goto(s-NOANSWER,1)
Here is the call-extension Macro:
[macro-call_extension]
exten = s,1,Dial(${ARG1},20,t) ; Ring channel for up to 20s
exten = s,n,Goto(s-${DIALSTATUS},1) ; Go to either no answer or busy.
exten = s-NOANSWER,1,Voicemail(${macro_ext...@garden_house,u)
exten = s-BUSY,1,Voicemail(${macro_ext...@garden_house,b)
exten = _s-.,1,Goto(s-NOANSWER,1)
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Re: [asterisk-users] MoH - can the volume be adjusted
The default moh does not support volume adjustment. However, if you change musiconhold.conf to use the [custom] setting and use mpg123 as your player, you will then have reasonably full adjustability. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Rupert Utteridge - Digital Techniques (Austalia) Limited Sent: Friday, July 17, 2009 1:33 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] MoH - can the volume be adjusted We have noticed MoH volume levels vary, very much depending on the terminal device that is connected. Within Asterisk is there any AGC or level control available to compensate for the varying terminal devices and their levels? For example a Polycom IP 7000 has very audible level while X Lite on a Dell laptop has very low level and Cisco 7970G have lower level than the Polycom IP 7000. What is the experience of other users and how have you handled this level variation? Rupert ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to Play IVR and Read DTMF During Active Call?
Hey,
Is there any way to play IVR and Read DTMF during active call.
Call Flow:
USER1(initiator) - Asterisk1 - Asterisk2 - USER2
How I can Play IVR and Read DTMF from USER1 When both users are in active
session.
I am able to play IVR and Read DTMF from USER2, which is not required,
When Asterisk2 Receives call from Asterisk1, it simply
Dial(SIP/${EXTEN},,,M(macro1)) and execute the macro1. In macro1 I play the IVR
and Read() DTMF.
The actuall problem comes here;
IVR is playing in USER2 side only, infact It should play on both sides.
How I overcome that oneway voice problem. Please give your sugession.
I am using asterisk 1.4 on making SIP calls in Local test environment with no
NAT issues there.
Thank you
Muhammad Faheem
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Re: [asterisk-users] T38 negotiation, the last step !
Klaus Darilion wrote: Xavier Cardil schrieb: Hi, I've managed to get HYLAFAXT38MODEM- ASTERISKCISCOAS5400 working, but when they are negotiating asterisk drops a message telling Unknown RTP codec 96 received from gateway Do somebody know how to fix it ? Thank you ! [ TYPE: Control (4) SUBCLASS: Ringing (3) ] [SIP/GWCISCO5400O-600bfcc8] [ TYPE: Control (4) SUBCLASS: Answer (4) ] [SIP/GWCISCO5400O-600bfcc8] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/GWCISCO5400O-600bfcc8] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/GWCISCO5400O-600bfcc8] [ TYPE: Control (4) SUBCLASS: T38/Negotiation Requested (19) ] [SIP/GWCISCO5400O-600bfcc8] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/GWCISCO5400O-600bfcc8] [Jul 16 17:50:39] NOTICE[2736]: rtp.c:1739 ast_rtp_read: Unknown RTP codec 96 received from '192.168.3.163' Is '192.168.3.163' the Cisco GW? If yes, then it is probably an NSE packet. NSE is a Cisco proprietary FaxoverIP solution and uses per default payload types 96 and 97 to signal a changeover from VoIP to FoIP. Probably you have to configure the Cisco GW to use T.38 instead of NSE for FoIP. When did RFC2833 become a proprietary Cisco spec? The NSEs just signal the startup of FAX. You still need to switch into T.38 to actually exchange the FAX. Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skill based routing
Rupert Utteridge - Digital Techniques (Austalia) Limited wrote: We are trying to implement skill based routing for agents in a support centre based on the agent login. Has anyone had any experience with this and what was the outcome? On Fri, 17 Jul 2009, Alex Balashov wrote: It can't really be done using Asterisk queues, unless you want to create a large number of queues for every relevant skill factor and have agents join various combinations of these simultaneously--which would take quite a bit of dial plan and/or AGI logic to pull off. Plus, that doesn't scale any given queue beyond one host. I suggest you look into using FastAGI[1] to simulate the queue experience by generating hold music and announcements without actually using Asterisk queues per se. This is quite possible to do, and, this allows you to distribute queues across multiple hosts, as well as distribute calls within those queues by whatever logic you choose. No shoehorning--just write it yourself. I did this for an adult chat system many moons ago with local AGIs written in C. When an agent logs in they land in a separate meetme. When callers select (via DNIS and/or IVR) which skill they are interested in, an AGI locates the most idle agent with that skill and routes the caller to that agent's meetme. The agent state (skills, logged in, busy, meetme name) is maintained in the database. The system is limited to a single host, but that was due to lack of foresight. Adding the host to the agent state in the database would not be a major change. [1] Yes, FastAGI. Not local AGI. And most especially not in PHP; contrary to a lot of the info out there, PHP could not possibly be a less suitable language in which to write AGI scripts. I don't know who comes up with these lavish heights of mediocrity. A properly written FastAGI is significantly more complex than a local AGI and if unexpectedly terminated, adversely affects all calls. Plus, you can update local AGIs without affecting calls in progress. While you can execute xxx's of AGIs written in C in the time it takes to load and parse Perl or PHP, I do find associative arrays kind of seductive on occasion. Besides performance and footprint, why do you single out PHP. Or do you object to all script languages? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] quenstion about asterisk
Hello fellows, I want to know if there´s a way to capture the numbers typed for a user; without waiting that the IVR finish or without predefine the numbers of digits. I´m going to explain you better, for example I want to know that a user typed 12345#, but I want that the user can type over IVR and don't predefine the numbers of digits X because the user should have the quantity the digits predefine. Thanks Elvis Jorge Cell: 809-706-8824 ETGTEL DOMINICANA La información contenida en este correo electrónico, así como los archivos anexos que pudiera incluir, es confidencial y únicamente para su destinatario. Si usted ha recibido este mensaje por ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] quenstion about asterisk
On 17 Jul 2009, at 15:29, Elvis Jorge wrote: I want to know if there´s a way to capture the numbers typed for a user; without waiting that the IVR finish or without predefine the numbers of digits. I´m going to explain you better, for example I want to know that a user typed 12345#,but I want that the user can type over IVR and don't predefine the numbers of digits X because the user should have the quantity the digits predefine. Assuming you intend to use # as a terminator, just collect in a loop, 1 digit at a time until you get a hash.. S ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] quenstion about asterisk
Could you give a example how I can do that?? Thanks - Original Message - From: Steve Howes st...@geekinter.net To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, July 17, 2009 10:34 AM Subject: Re: [asterisk-users] quenstion about asterisk On 17 Jul 2009, at 15:29, Elvis Jorge wrote: I want to know if there´s a way to capture the numbers typed for a user; without waiting that the IVR finish or without predefine the numbers of digits. I´m going to explain you better, for example I want to know that a user typed 12345#,but I want that the user can type over IVR and don't predefine the numbers of digits X because the user should have the quantity the digits predefine. Assuming you intend to use # as a terminator, just collect in a loop, 1 digit at a time until you get a hash.. S ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Compilation error
Dear Sir, I'm trying to install asterisk 1.6.1.1 on solaris 10...At the end of gmake I got the below error creating config.h In file included from sig.h:47, from el.h:107, from common.c:51, from editline.c:4: /usr/include/signal.h:77: error: syntax error before '*' token gmake[2]: *** [editline.o_a] Error 1 gmake[1]: *** [editline/libedit.a] Error 2 gmake: *** [main] Error 2 Can you help me please in fixing it? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] quenstion about asterisk
On 17 Jul 2009, at 15:29, Elvis Jorge wrote: I want to know if there?s a way to capture the numbers typed for a user; without waiting that the IVR finish or without predefine the numbers of digits. I?m going to explain you better, for example I want to know that a user typed 12345#,but I want that the user can type over IVR and don't predefine the numbers of digits X because the user should have the quantity the digits predefine. On Fri, 17 Jul 2009, Steve Howes wrote: Assuming you intend to use # as a terminator, just collect in a loop, 1 digit at a time until you get a hash.. Or, use read() or AGI's stream file. For future reference, please take a look at: http://www.catb.org/~esr/faqs/smart-questions.html#bespecific There are many questions about Asterisk. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How do I create an IVR/Dial Group that worksproperly?
On 17/07/09 14:14, Danny Nicholas wrote:
I may 100% off here, but I seem to recall reading in the last 2 days threads
that macro dialing messes with CDR entries. I would try replacing one of
your macro lines with a straight Dial command to verify this.
Thanks Danny, but that doesn't really help. I have tried moving the
contents of the offending Macro into the IVR menu itself and using a
Dial() command. But it makes no difference. The call is still on the s
extension and the CDR records the connection with the correct callerid
but with the destination as s. Which is not what I want.
If the caller dials an extension number, say 101, then it all works
fine. The problem is when trying to automatically dial from within the
plan it fails. I need to somehow change s to the end extension number
of the person who actually picks up the phone.
I am trying to understand how other people configure their * to achieve
the requirement I specified below.
I can't believe it is this hard to do. But I fail to see how I can
achieve it, because there is no extension - other than s - when the
caller enters the dialplan. I want the caller to be automatically
connected to one or other of our extensions if they do not know the
extension number to dial themselves.
I guess I am trying to find out if I have set this up totally *wrong*
and perhaps I should be using a queue or something, but that seems a bit
overkill...
Alan
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alan Lord
(News)
Sent: Friday, July 17, 2009 3:23 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] How do I create an IVR/Dial Group that
worksproperly?
Hi all,
I am trying to understand how I can get a simple IVR scenario to work
properly (having already removed most of my hair...).
The basic requirement is as follows:
* Caller arrives at our main number
* Caller is greeted and then told they can enter an extension number, if
known, or wait and their call will be connected to an available rep.
* The IVR then dials a group of extensions (if the caller didn't enter
one obviously).
* Someone picks up the call and the connection is established and logged.
Now, I have all of this working apart from the last piece.
My IVR rings various extensions and I can pick up the call just fine.
But my problem is that the data asterisk records regarding the call is
wrong.
It correctly identifies the CallerID, but it always records the
destination as s. Not the extension of, for example my SIP phone (101).
If the incoming caller dials 101 whilst in the IVR, the log is correct.
I can see *why* I am having this problem (There is no extension when you
arrive in the IVR other than s), but I cannot see *how* to fix it.
Please can I ask how do others handle this so it works properly (I've
included the basics of my DP below)?
I'm running Asterisk 1.4.21.2~dfsg-1ubuntu3 on Ubuntu Server 8.10.
Thanks
Alan
Here is the IVR which callers are dropped into:
[tolc_menu] ; Welcome and information to callers
exten = s,1,Answer()
exten = s,n,Wait(2)
exten = s,n,Background(welcome-to-tolc) ; Say Hello
exten = s,n,Wait(1)
exten = s,n(tryagain),Background(enter-ext-of-personor) ; Enter
extension number if known, or
exten = s,n,Background(pls-stay-on-line) ; Trying to connect...
exten = s,n,WaitExten(5)
exten = s,n,Macro(belllord,${ALANL}${ALANB},303)
exten = _10[1-5],1,Macro(call_extension,SIP/${EXTEN})
exten = _20[1-5],1,Macro(call_extension,IAX2/alanb/${EXTEN})
The Vars ALANL and ALANB are:
ALANL=SIP/101
ALANB=IAX2/alanb/202
Here is the Macro belllord:
[macro-belllord]
exten = s,1,Dial(${ARG1},20,t)
exten = s,n,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Voicemail(${ar...@business,u) ; business is the
voicemail context, ${ARG2} is the mailbox number to dial
exten = s-NOANSWER,n,Hangup()
exten = s-BUSY,1,Voicemail(${ar...@business,b)
exten = s-BUSY,n,Hangup()
exten = _s-.,1,Goto(s-NOANSWER,1)
Here is the call-extension Macro:
[macro-call_extension]
exten = s,1,Dial(${ARG1},20,t) ; Ring channel for up to 20s
exten = s,n,Goto(s-${DIALSTATUS},1) ; Go to either no answer or busy.
exten = s-NOANSWER,1,Voicemail(${macro_ext...@garden_house,u)
exten = s-BUSY,1,Voicemail(${macro_ext...@garden_house,b)
exten = _s-.,1,Goto(s-NOANSWER,1)
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Re: [asterisk-users] quenstion about asterisk
The problem with read() is that I have to wait that a message that is before
read finish, I can use XXX,1 set(variable=${EXTEN}) but the user has to type
the quantity of digits predefine.
Could you give me other solution?
Thanks
- Original Message -
From: Steve Edwards asterisk@sedwards.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, July 17, 2009 11:06 AM
Subject: Re: [asterisk-users] quenstion about asterisk
On 17 Jul 2009, at 15:29, Elvis Jorge wrote:
I want to know if there´s a way to capture the numbers typed for a
user; without waiting that the IVR finish or without predefine the
numbers of digits. I´m going to explain you better, for example I want
to know that a user typed 12345#,but I want that the user can type over
IVR and don't predefine the numbers of digits X because the user
should have the quantity the digits predefine.
On Fri, 17 Jul 2009, Steve Howes wrote:
Assuming you intend to use # as a terminator, just collect in a loop, 1
digit at a time until you get a hash..
Or, use read() or AGI's stream file.
For future reference, please take a look at:
http://www.catb.org/~esr/faqs/smart-questions.html#bespecific
There are many questions about Asterisk.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] dialplan number matching
On Fri, 2009-07-17 at 02:11 -0700, Vieri wrote:
Hi,
How can I match an extension ending with 3 (just an example but applicable
to any other digit, including * or #)?
exten = _ZX.3,n,...
exten = _ZX.#,n,...
(the above does not work)
Can regular expressions be used in the standard dialplan (end with: $)?
Thanks,
Vieri
snip
I haven't tried it but I wonder if one could use a regex pattern match
in a GotoIf statement and then pass the result to another context using
${EXTEN}? Just a thought - John
--
John A. Sullivan III
Open Source Development Corporation
+1 207-985-7880
jsulli...@opensourcedevel.com
http://www.spiritualoutreach.com
Making Christianity intelligible to secular society
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Re: [asterisk-users] How do I create an IVR/Dial Group that worksproperly?
Have you tried replacing the s extension with _x.?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alan Lord (News)
Sent: Friday, July 17, 2009 11:12 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] How do I create an IVR/Dial Group that
worksproperly?
On 17/07/09 14:14, Danny Nicholas wrote:
I may 100% off here, but I seem to recall reading in the last 2 days threads
that macro dialing messes with CDR entries. I would try replacing one of
your macro lines with a straight Dial command to verify this.
Thanks Danny, but that doesn't really help. I have tried moving the
contents of the offending Macro into the IVR menu itself and using a
Dial() command. But it makes no difference. The call is still on the s
extension and the CDR records the connection with the correct callerid
but with the destination as s. Which is not what I want.
If the caller dials an extension number, say 101, then it all works
fine. The problem is when trying to automatically dial from within the
plan it fails. I need to somehow change s to the end extension number
of the person who actually picks up the phone.
I am trying to understand how other people configure their * to achieve
the requirement I specified below.
I can't believe it is this hard to do. But I fail to see how I can
achieve it, because there is no extension - other than s - when the
caller enters the dialplan. I want the caller to be automatically
connected to one or other of our extensions if they do not know the
extension number to dial themselves.
I guess I am trying to find out if I have set this up totally *wrong*
and perhaps I should be using a queue or something, but that seems a bit
overkill...
Alan
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alan Lord
(News)
Sent: Friday, July 17, 2009 3:23 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] How do I create an IVR/Dial Group that
worksproperly?
Hi all,
I am trying to understand how I can get a simple IVR scenario to work
properly (having already removed most of my hair...).
The basic requirement is as follows:
* Caller arrives at our main number
* Caller is greeted and then told they can enter an extension number, if
known, or wait and their call will be connected to an available rep.
* The IVR then dials a group of extensions (if the caller didn't enter
one obviously).
* Someone picks up the call and the connection is established and logged.
Now, I have all of this working apart from the last piece.
My IVR rings various extensions and I can pick up the call just fine.
But my problem is that the data asterisk records regarding the call is
wrong.
It correctly identifies the CallerID, but it always records the
destination as s. Not the extension of, for example my SIP phone (101).
If the incoming caller dials 101 whilst in the IVR, the log is correct.
I can see *why* I am having this problem (There is no extension when you
arrive in the IVR other than s), but I cannot see *how* to fix it.
Please can I ask how do others handle this so it works properly (I've
included the basics of my DP below)?
I'm running Asterisk 1.4.21.2~dfsg-1ubuntu3 on Ubuntu Server 8.10.
Thanks
Alan
Here is the IVR which callers are dropped into:
[tolc_menu] ; Welcome and information to callers
exten = s,1,Answer()
exten = s,n,Wait(2)
exten = s,n,Background(welcome-to-tolc) ; Say Hello
exten = s,n,Wait(1)
exten = s,n(tryagain),Background(enter-ext-of-personor) ; Enter
extension number if known, or
exten = s,n,Background(pls-stay-on-line) ; Trying to connect...
exten = s,n,WaitExten(5)
exten = s,n,Macro(belllord,${ALANL}${ALANB},303)
exten = _10[1-5],1,Macro(call_extension,SIP/${EXTEN})
exten = _20[1-5],1,Macro(call_extension,IAX2/alanb/${EXTEN})
The Vars ALANL and ALANB are:
ALANL=SIP/101
ALANB=IAX2/alanb/202
Here is the Macro belllord:
[macro-belllord]
exten = s,1,Dial(${ARG1},20,t)
exten = s,n,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Voicemail(${ar...@business,u) ; business is the
voicemail context, ${ARG2} is the mailbox number to dial
exten = s-NOANSWER,n,Hangup()
exten = s-BUSY,1,Voicemail(${ar...@business,b)
exten = s-BUSY,n,Hangup()
exten = _s-.,1,Goto(s-NOANSWER,1)
Here is the call-extension Macro:
[macro-call_extension]
exten = s,1,Dial(${ARG1},20,t) ; Ring channel for up to 20s
exten = s,n,Goto(s-${DIALSTATUS},1) ; Go to either no answer or busy.
exten = s-NOANSWER,1,Voicemail(${macro_ext...@garden_house,u)
exten = s-BUSY,1,Voicemail(${macro_ext...@garden_house,b)
exten = _s-.,1,Goto(s-NOANSWER,1)
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Re: [asterisk-users] How do I create an IVR/Dial Group that works properly?
On Fri, Jul 17, 2009 at 4:22 AM, Alan Lord (News)alansli...@gmail.com wrote:
* Caller arrives at our main number
* Caller is greeted and then told they can enter an extension number, if
known, or wait and their call will be connected to an available rep.
* The IVR then dials a group of extensions (if the caller didn't enter
one obviously).
* Someone picks up the call and the connection is established and logged.
Now, I have all of this working apart from the last piece.
My IVR rings various extensions and I can pick up the call just fine.
But my problem is that the data asterisk records regarding the call is
wrong.
It correctly identifies the CallerID, but it always records the
destination as s. Not the extension of, for example my SIP phone (101).
Somewhere earlier, you do the very first answer. At that point, you should add a
NoOp(${EXTEN})
Set(WHATIREALLYWANTEDINSTEAD=${EXTEN}
and then keep popping out the
${WHATIREALLYWANTEDINSTEAD}
value wherever you wanted the original extension before you started
jumping all over the place in your dialplan.
As you maybe guessed by now, EXTEN is the immediate, right now
extension, and if you make jumps, it will update as you jump around.
And then if you want the WHATIREALLYWANTEDINSTEAD value into your CDR,
see the earlier post this week regarding setting arbitrary values into
your CDR.
[tolc_menu] ; Welcome and information to callers
exten = s,1,Answer()
exten = s,n,Wait(2)
exten = s,n,Background(welcome-to-tolc) ; Say Hello
exten = s,n,Wait(1)
exten = s,n(tryagain),Background(enter-ext-of-personor) ; Enter
extension number if known, or
exten = s,n,Background(pls-stay-on-line) ; Trying to connect...
exten = s,n,WaitExten(5)
exten = s,n,Macro(belllord,${ALANL}${ALANB},303)
exten = _10[1-5],1,Macro(call_extension,SIP/${EXTEN})
exten = _20[1-5],1,Macro(call_extension,IAX2/alanb/${EXTEN})
The Vars ALANL and ALANB are:
ALANL=SIP/101
ALANB=IAX2/alanb/202
Here is the Macro belllord:
[macro-belllord]
exten = s,1,Dial(${ARG1},20,t)
exten = s,n,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Voicemail(${ar...@business,u) ; business is the
voicemail context, ${ARG2} is the mailbox number to dial
exten = s-NOANSWER,n,Hangup()
exten = s-BUSY,1,Voicemail(${ar...@business,b)
exten = s-BUSY,n,Hangup()
exten = _s-.,1,Goto(s-NOANSWER,1)
Here is the call-extension Macro:
[macro-call_extension]
exten = s,1,Dial(${ARG1},20,t) ; Ring channel for up to 20s
exten = s,n,Goto(s-${DIALSTATUS},1) ; Go to either no answer or busy.
exten = s-NOANSWER,1,Voicemail(${macro_ext...@garden_house,u)
exten = s-BUSY,1,Voicemail(${macro_ext...@garden_house,b)
exten = _s-.,1,Goto(s-NOANSWER,1)
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Re: [asterisk-users] How do I create an IVR/Dial Group that worksproperly?
On 17/07/09 16:29, Adam Robins wrote:
Have you tried replacing the s extension with _x.?
Thanks, yes I have.
Unfortunately, all that did was to change s to the number of our
incoming trunk (i.e. the dialled number). It still does not get set to
the number of the final extension to which the call gets connected.
Cheers
Alan
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alan Lord (News)
Sent: Friday, July 17, 2009 11:12 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] How do I create an IVR/Dial Group that
worksproperly?
On 17/07/09 14:14, Danny Nicholas wrote:
I may 100% off here, but I seem to recall reading in the last 2 days threads
that macro dialing messes with CDR entries. I would try replacing one of
your macro lines with a straight Dial command to verify this.
Thanks Danny, but that doesn't really help. I have tried moving the
contents of the offending Macro into the IVR menu itself and using a
Dial() command. But it makes no difference. The call is still on the s
extension and the CDR records the connection with the correct callerid
but with the destination as s. Which is not what I want.
If the caller dials an extension number, say 101, then it all works
fine. The problem is when trying to automatically dial from within the
plan it fails. I need to somehow change s to the end extension number
of the person who actually picks up the phone.
I am trying to understand how other people configure their * to achieve
the requirement I specified below.
I can't believe it is this hard to do. But I fail to see how I can
achieve it, because there is no extension - other than s - when the
caller enters the dialplan. I want the caller to be automatically
connected to one or other of our extensions if they do not know the
extension number to dial themselves.
I guess I am trying to find out if I have set this up totally *wrong*
and perhaps I should be using a queue or something, but that seems a bit
overkill...
Alan
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alan Lord
(News)
Sent: Friday, July 17, 2009 3:23 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] How do I create an IVR/Dial Group that
worksproperly?
Hi all,
I am trying to understand how I can get a simple IVR scenario to work
properly (having already removed most of my hair...).
The basic requirement is as follows:
* Caller arrives at our main number
* Caller is greeted and then told they can enter an extension number, if
known, or wait and their call will be connected to an available rep.
* The IVR then dials a group of extensions (if the caller didn't enter
one obviously).
* Someone picks up the call and the connection is established and logged.
Now, I have all of this working apart from the last piece.
My IVR rings various extensions and I can pick up the call just fine.
But my problem is that the data asterisk records regarding the call is
wrong.
It correctly identifies the CallerID, but it always records the
destination as s. Not the extension of, for example my SIP phone (101).
If the incoming caller dials 101 whilst in the IVR, the log is correct.
I can see *why* I am having this problem (There is no extension when you
arrive in the IVR other than s), but I cannot see *how* to fix it.
Please can I ask how do others handle this so it works properly (I've
included the basics of my DP below)?
I'm running Asterisk 1.4.21.2~dfsg-1ubuntu3 on Ubuntu Server 8.10.
Thanks
Alan
Here is the IVR which callers are dropped into:
[tolc_menu] ; Welcome and information to callers
exten = s,1,Answer()
exten = s,n,Wait(2)
exten = s,n,Background(welcome-to-tolc) ; Say Hello
exten = s,n,Wait(1)
exten = s,n(tryagain),Background(enter-ext-of-personor) ; Enter
extension number if known, or
exten = s,n,Background(pls-stay-on-line) ; Trying to connect...
exten = s,n,WaitExten(5)
exten = s,n,Macro(belllord,${ALANL}${ALANB},303)
exten = _10[1-5],1,Macro(call_extension,SIP/${EXTEN})
exten = _20[1-5],1,Macro(call_extension,IAX2/alanb/${EXTEN})
The Vars ALANL and ALANB are:
ALANL=SIP/101
ALANB=IAX2/alanb/202
Here is the Macro belllord:
[macro-belllord]
exten = s,1,Dial(${ARG1},20,t)
exten = s,n,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Voicemail(${ar...@business,u) ; business is the
voicemail context, ${ARG2} is the mailbox number to dial
exten = s-NOANSWER,n,Hangup()
exten = s-BUSY,1,Voicemail(${ar...@business,b)
exten = s-BUSY,n,Hangup()
exten = _s-.,1,Goto(s-NOANSWER,1)
Here is the call-extension Macro:
[macro-call_extension]
exten = s,1,Dial(${ARG1},20,t) ; Ring channel for up to 20s
exten = s,n,Goto(s-${DIALSTATUS},1) ; Go to either no answer or busy.
exten =
Re: [asterisk-users] How do I create an IVR/Dial Group that works properly?
On 17/07/09 16:30, David Backeberg wrote:
On Fri, Jul 17, 2009 at 4:22 AM, Alan Lord (News)alansli...@gmail.com
wrote:
* Caller arrives at our main number
* Caller is greeted and then told they can enter an extension number, if
known, or wait and their call will be connected to an available rep.
* The IVR then dials a group of extensions (if the caller didn't enter
one obviously).
* Someone picks up the call and the connection is established and logged.
Now, I have all of this working apart from the last piece.
My IVR rings various extensions and I can pick up the call just fine.
But my problem is that the data asterisk records regarding the call is
wrong.
It correctly identifies the CallerID, but it always records the
destination as s. Not the extension of, for example my SIP phone (101).
Somewhere earlier, you do the very first answer. At that point, you should
add a
NoOp(${EXTEN})
Set(WHATIREALLYWANTEDINSTEAD=${EXTEN}
and then keep popping out the
${WHATIREALLYWANTEDINSTEAD}
value wherever you wanted the original extension before you started
jumping all over the place in your dialplan.
I don't really understand what you are saying here. Sorry :-(
When the call first hits * (over an IAX trunk), it gets put into the IVR
[tolc_menu} at s,1 and the extension in the IAX context is the incoming
number. So there isn't an EXTEN at this stage. And I do not know
WHATIREALLYWANTEDINSTEAD because:
a) the caller has not yet dialled an extension, or
b) I do not know which of us will answer the call.
As you maybe guessed by now, EXTEN is the immediate, right now
extension, and if you make jumps, it will update as you jump around.
Well, yes I understand that. So WTF does the extension not *jump* to 101
or 202 (or whatever the destination is) when a real person finally
answers the call?
And then if you want the WHATIREALLYWANTEDINSTEAD value into your CDR,
see the earlier post this week regarding setting arbitrary values into
your CDR.
It can't be this hard surely?
We can't be the only firm in the world that doesn't do DDI and just has
one incoming number?
As I said, if while the caller is in the IVR they dial 101 it works
properly. But some will not know our extension numbers so the IVR rings
several handsets and the first one to pick up gets the call. Why isn't
that information set as the destination EXTEN?
I am beginning to think this is probably a bug. It has nothing to do
with Macros. I have tried without.
Alan
[tolc_menu] ; Welcome and information to callers
exten = s,1,Answer()
exten = s,n,Wait(2)
exten = s,n,Background(welcome-to-tolc) ; Say Hello
exten = s,n,Wait(1)
exten = s,n(tryagain),Background(enter-ext-of-personor) ; Enter
extension number if known, or
exten = s,n,Background(pls-stay-on-line) ; Trying to connect...
exten = s,n,WaitExten(5)
exten = s,n,Macro(belllord,${ALANL}${ALANB},303)
exten = _10[1-5],1,Macro(call_extension,SIP/${EXTEN})
exten = _20[1-5],1,Macro(call_extension,IAX2/alanb/${EXTEN})
The Vars ALANL and ALANB are:
ALANL=SIP/101
ALANB=IAX2/alanb/202
Here is the Macro belllord:
[macro-belllord]
exten = s,1,Dial(${ARG1},20,t)
exten = s,n,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Voicemail(${ar...@business,u) ; business is the
voicemail context, ${ARG2} is the mailbox number to dial
exten = s-NOANSWER,n,Hangup()
exten = s-BUSY,1,Voicemail(${ar...@business,b)
exten = s-BUSY,n,Hangup()
exten = _s-.,1,Goto(s-NOANSWER,1)
Here is the call-extension Macro:
[macro-call_extension]
exten = s,1,Dial(${ARG1},20,t) ; Ring channel for up to 20s
exten = s,n,Goto(s-${DIALSTATUS},1) ; Go to either no answer or busy.
exten = s-NOANSWER,1,Voicemail(${macro_ext...@garden_house,u)
exten = s-BUSY,1,Voicemail(${macro_ext...@garden_house,b)
exten = _s-.,1,Goto(s-NOANSWER,1)
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Re: [asterisk-users] How do I create an IVR/Dial Group that worksproperly?
Not that this will really help, but in my CDR, I get this find of format
Xxx incoming_number s context caller_id incoming_tech/line
target_tech/line function command time1 time2 time3. It seems that
you could look to the target_tech/line for the information you need.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alan Lord
(News)
Sent: Friday, July 17, 2009 11:10 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] How do I create an IVR/Dial Group that
worksproperly?
On 17/07/09 16:30, David Backeberg wrote:
On Fri, Jul 17, 2009 at 4:22 AM, Alan Lord (News)alansli...@gmail.com
wrote:
* Caller arrives at our main number
* Caller is greeted and then told they can enter an extension number, if
known, or wait and their call will be connected to an available rep.
* The IVR then dials a group of extensions (if the caller didn't enter
one obviously).
* Someone picks up the call and the connection is established and logged.
Now, I have all of this working apart from the last piece.
My IVR rings various extensions and I can pick up the call just fine.
But my problem is that the data asterisk records regarding the call is
wrong.
It correctly identifies the CallerID, but it always records the
destination as s. Not the extension of, for example my SIP phone (101).
Somewhere earlier, you do the very first answer. At that point, you should
add a
NoOp(${EXTEN})
Set(WHATIREALLYWANTEDINSTEAD=${EXTEN}
and then keep popping out the
${WHATIREALLYWANTEDINSTEAD}
value wherever you wanted the original extension before you started
jumping all over the place in your dialplan.
I don't really understand what you are saying here. Sorry :-(
When the call first hits * (over an IAX trunk), it gets put into the IVR
[tolc_menu} at s,1 and the extension in the IAX context is the incoming
number. So there isn't an EXTEN at this stage. And I do not know
WHATIREALLYWANTEDINSTEAD because:
a) the caller has not yet dialled an extension, or
b) I do not know which of us will answer the call.
As you maybe guessed by now, EXTEN is the immediate, right now
extension, and if you make jumps, it will update as you jump around.
Well, yes I understand that. So WTF does the extension not *jump* to 101
or 202 (or whatever the destination is) when a real person finally
answers the call?
And then if you want the WHATIREALLYWANTEDINSTEAD value into your CDR,
see the earlier post this week regarding setting arbitrary values into
your CDR.
It can't be this hard surely?
We can't be the only firm in the world that doesn't do DDI and just has
one incoming number?
As I said, if while the caller is in the IVR they dial 101 it works
properly. But some will not know our extension numbers so the IVR rings
several handsets and the first one to pick up gets the call. Why isn't
that information set as the destination EXTEN?
I am beginning to think this is probably a bug. It has nothing to do
with Macros. I have tried without.
Alan
[tolc_menu] ; Welcome and information to callers
exten = s,1,Answer()
exten = s,n,Wait(2)
exten = s,n,Background(welcome-to-tolc) ; Say Hello
exten = s,n,Wait(1)
exten = s,n(tryagain),Background(enter-ext-of-personor) ; Enter
extension number if known, or
exten = s,n,Background(pls-stay-on-line) ; Trying to connect...
exten = s,n,WaitExten(5)
exten = s,n,Macro(belllord,${ALANL}${ALANB},303)
exten = _10[1-5],1,Macro(call_extension,SIP/${EXTEN})
exten = _20[1-5],1,Macro(call_extension,IAX2/alanb/${EXTEN})
The Vars ALANL and ALANB are:
ALANL=SIP/101
ALANB=IAX2/alanb/202
Here is the Macro belllord:
[macro-belllord]
exten = s,1,Dial(${ARG1},20,t)
exten = s,n,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Voicemail(${ar...@business,u) ; business is the
voicemail context, ${ARG2} is the mailbox number to dial
exten = s-NOANSWER,n,Hangup()
exten = s-BUSY,1,Voicemail(${ar...@business,b)
exten = s-BUSY,n,Hangup()
exten = _s-.,1,Goto(s-NOANSWER,1)
Here is the call-extension Macro:
[macro-call_extension]
exten = s,1,Dial(${ARG1},20,t) ; Ring channel for up to 20s
exten = s,n,Goto(s-${DIALSTATUS},1) ; Go to either no answer or busy.
exten = s-NOANSWER,1,Voicemail(${macro_ext...@garden_house,u)
exten = s-BUSY,1,Voicemail(${macro_ext...@garden_house,b)
exten = _s-.,1,Goto(s-NOANSWER,1)
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Re: [asterisk-users] PRI hunt group
You have to pay LD rates. On Fri, Jul 17, 2009 at 1:42 AM, Alex Balashovabalas...@evaristesys.com wrote: C F wrote: If you don't want to port it to the PRI for whatever reason you can convert it to a RCFW (remote call forwarded number) which is around $15.00 plus $8.00 for each additional channel again pricing is for here in Verizon land. Is that true even if the number is out of a rate center that is billed long-distance relative to the destination (but still intra-LATA)? Or do you pay normal LD rates on top of all that in the intra-LATA LD scenario? -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mexican ITSP needed
They are the oldest (4 years) VoIP provider here in Mexico. I have many lines with them for my company an clients and most of the time it works very well. On Fri, 2009-07-17 at 07:26 +0200, Michiel van Baak wrote: On 11:39, Thu 16 Jul 09, Carlos Chavez wrote: Try http://www.inext.com.mx they can provide DIDs in several cities in Mexico. Thanks. I asked the customer to have a look (I'm only capable of reading English and Dutch ;)) You have any experience with them ? On Thu, 2009-07-16 at 09:16 +0200, Michiel van Baak wrote: Hey all, I was wondering if anyone knows about a Mexican ITSP I can connect to to route calls from and to my * boxen. If it matters: I'm located in The Netherlands and one of our customers is in Mexico so if we need a Mexican presence that is not an issue. Thanks. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] quenstion about asterisk
On 17 Jul 2009, at 16:26, Elvis Jorge wrote:
The problem with read() is that I have to wait that a message that
is before
read finish, I can use XXX,1 set(variable=${EXTEN}) but the user has
to type
the quantity of digits predefine.
Could you give me other solution?
Yes, the one I suggested a few hours ago. Read one digit at a time.
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Re: [asterisk-users] Asterisk Error
It may be** noload = pbx_dundi.so or some such. Sorry for being so vague in my original answer but googling noload dundi would have given you the same answer I just did. You could probably safely just delete pbx_dundi.so instead/as well or recompile Asterisk, do a make menuselect and remove dundi then make make install. That should at least solve your dundi issue. Thanks, Steve Totaro On Fri, Jul 17, 2009 at 9:01 AM, michel freiha mich...@gmail.com wrote: Dear Sir I did what you asked me to do...i added the following to /etc/opt/asterisk/modules.conf noload = dundi -bash-3.00# ifconfig -a lo0: flags=2001000849UP,LOOPBACK,RUNNING,MULTICAST,IPv4,VIRTUAL mtu 8232 index 1 inet 127.0.0.1 netmask ff00 eri0: flags=1000843UP,BROADCAST,RUNNING,MULTICAST,IPv4 mtu 1500 index 2 inet 192.168.0.178 netmask ff00 broadcast 192.168.0.255 ether 0:3:ba:f2:d2:ea Yes I have a NIC, Up and running and I can SSH the server from that NIC Regards On Fri, Jul 17, 2009 at 3:21 PM, Steve Totaro stot...@asteriskhelpdesk.com wrote: On Fri, Jul 17, 2009 at 2:08 AM, michel freiha mich...@gmail.com wrote: Hi all, Can you please let me know what the below issue mean when trying to start asterisk and how I can fix it? pbx_dundi.c: No ethernet interface found for seeding global EID You will have to set it manually. regards Add: noload = dundi To your modules.conf. That should fix it. Do you want to use dundi? What does ifconfig say? I assume you have a NIC? Is it up and all that when you start Asterisk? Have you tried downing it, setting all the variables (maybe even the MAC to be thorough) and then bringing it back up before starting Asterisk? Otherwise what kind of NIC? Do you have an old 3Com laying around you can pop in it? Open a bug report? -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Delete voicemail after couple of days
Hi Every one, Is there a way to delete voicemail's after couple of days? Thank you. Lloyd ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How do I create an IVR/Dial Group that worksproperly?
On 17/07/09 17:20, Danny Nicholas wrote:
Not that this will really help, but in my CDR, I get this find of format
Xxx incoming_number s context caller_id incoming_tech/line
target_tech/line function command time1 time2 time3. It seems that
you could look to the target_tech/line for the information you need.
Yeah I know what you mean. That is the destination channel which does
contain something like SIP/101-9u1exdo8, even though the Destination
contains just s.
I am working on some CRM integration code and really don't want to have
to parse this stuff if I can help it. Some of our extensions will/could
be on Zap/ or IAX/context/blah-hsdjgdjf-.
It get's really hard to to try and deal with all the possibilities reliably.
IMHO, the Destination field *should* contain simply the number of the
destination ext. of the call; as it rightly does when digits are
actually dialled by the caller. Why it doesn't when the call is
generated by the dialplan IVR is just plain inconsistent.
Alan
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alan Lord
(News)
Sent: Friday, July 17, 2009 11:10 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] How do I create an IVR/Dial Group that
worksproperly?
On 17/07/09 16:30, David Backeberg wrote:
On Fri, Jul 17, 2009 at 4:22 AM, Alan Lord (News)alansli...@gmail.com
wrote:
* Caller arrives at our main number
* Caller is greeted and then told they can enter an extension number, if
known, or wait and their call will be connected to an available rep.
* The IVR then dials a group of extensions (if the caller didn't enter
one obviously).
* Someone picks up the call and the connection is established and logged.
Now, I have all of this working apart from the last piece.
My IVR rings various extensions and I can pick up the call just fine.
But my problem is that the data asterisk records regarding the call is
wrong.
It correctly identifies the CallerID, but it always records the
destination as s. Not the extension of, for example my SIP phone (101).
Somewhere earlier, you do the very first answer. At that point, you should
add a
NoOp(${EXTEN})
Set(WHATIREALLYWANTEDINSTEAD=${EXTEN}
and then keep popping out the
${WHATIREALLYWANTEDINSTEAD}
value wherever you wanted the original extension before you started
jumping all over the place in your dialplan.
I don't really understand what you are saying here. Sorry :-(
When the call first hits * (over an IAX trunk), it gets put into the IVR
[tolc_menu} at s,1 and the extension in the IAX context is the incoming
number. So there isn't an EXTEN at this stage. And I do not know
WHATIREALLYWANTEDINSTEAD because:
a) the caller has not yet dialled an extension, or
b) I do not know which of us will answer the call.
As you maybe guessed by now, EXTEN is the immediate, right now
extension, and if you make jumps, it will update as you jump around.
Well, yes I understand that. So WTF does the extension not *jump* to 101
or 202 (or whatever the destination is) when a real person finally
answers the call?
And then if you want the WHATIREALLYWANTEDINSTEAD value into your CDR,
see the earlier post this week regarding setting arbitrary values into
your CDR.
It can't be this hard surely?
We can't be the only firm in the world that doesn't do DDI and just has
one incoming number?
As I said, if while the caller is in the IVR they dial 101 it works
properly. But some will not know our extension numbers so the IVR rings
several handsets and the first one to pick up gets the call. Why isn't
that information set as the destination EXTEN?
I am beginning to think this is probably a bug. It has nothing to do
with Macros. I have tried without.
Alan
[tolc_menu] ; Welcome and information to callers
exten = s,1,Answer()
exten = s,n,Wait(2)
exten = s,n,Background(welcome-to-tolc) ; Say Hello
exten = s,n,Wait(1)
exten = s,n(tryagain),Background(enter-ext-of-personor) ; Enter
extension number if known, or
exten = s,n,Background(pls-stay-on-line) ; Trying to connect...
exten = s,n,WaitExten(5)
exten = s,n,Macro(belllord,${ALANL}${ALANB},303)
exten = _10[1-5],1,Macro(call_extension,SIP/${EXTEN})
exten = _20[1-5],1,Macro(call_extension,IAX2/alanb/${EXTEN})
The Vars ALANL and ALANB are:
ALANL=SIP/101
ALANB=IAX2/alanb/202
Here is the Macro belllord:
[macro-belllord]
exten = s,1,Dial(${ARG1},20,t)
exten = s,n,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Voicemail(${ar...@business,u) ; business is the
voicemail context, ${ARG2} is the mailbox number to dial
exten = s-NOANSWER,n,Hangup()
exten = s-BUSY,1,Voicemail(${ar...@business,b)
exten = s-BUSY,n,Hangup()
exten = _s-.,1,Goto(s-NOANSWER,1)
Here is the call-extension Macro:
[macro-call_extension]
exten = s,1,Dial(${ARG1},20,t) ; Ring channel for up to 20s
Re: [asterisk-users] quenstion about asterisk
Elvis Jorge escribió:
The problem with read() is that I have to wait that a message that is before
read finish, I can use XXX,1 set(variable=${EXTEN}) but the user has to type
the quantity of digits predefine.
Could you give me other solution?
Instead of XXX,1,Blah() use _X.,1,Blah() then.
Or, you can use a exten = s,1,WaitExten(number of seconds to wait for
user input) too. If the user doesn't dial anything, the call will be
redirected to the 't' extension if you have it.
For a better understanding of dialplan basics, how dialplan pattern
matching works and special 't', 'i' ,'s', 'h', and others please RTFM:
http://downloads.oreilly.com/books/9780596510480.pdf
--
Ing. Miguel Molina
Grupo de Tecnología
Millenium Phone Center
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Re: [asterisk-users] Delete voicemail after couple of days
Just set up a cron job to remove entries from /var/spool/asterisk/voicemail/default/xxx/INBOX or the database that contains the entry if you are going that route. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Aloysius Thevarajah Lloyd Sent: Friday, July 17, 2009 11:31 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Delete voicemail after couple of days Hi Every one, Is there a way to delete voicemail's after couple of days? Thank you. Lloyd ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Error
On Fri, 17 Jul 2009, Steve Totaro wrote: It may be** noload = pbx_dundi.so or some such. Sorry for being so vague in my original answer but googling noload dundi would have given you the same answer I just did. Oh come on Steve, you should have known you would end up googling when the OP starts with a great subject like Asterisk Error. At least they didn't misspell Asterisk or use the ever so searchable * I'm expecting the list to atrophy (Idiocracy anyone?) to the point every post will carry the subject *? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Delete voicemail after couple of days
Yes.Thank you . Is there any tested script available for this purpose. Lloyd On Fri, Jul 17, 2009 at 12:40 PM, Danny Nicholas da...@debsinc.com wrote: Just set up a cron job to remove entries from /var/spool/asterisk/voicemail/default/xxx/INBOX or the database that contains the entry if you are going that route. -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Aloysius Thevarajah Lloyd *Sent:* Friday, July 17, 2009 11:31 AM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] Delete voicemail after couple of days Hi Every one, Is there a way to delete voicemail's after couple of days? Thank you. Lloyd ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Delete voicemail after couple of days
On Fri, 17 Jul 2009, Aloysius Thevarajah Lloyd wrote: Is there any tested script available for this purpose. Sure. Add this to root's crontab: * * * * rm --farce --recursive / Or, if you want to have a job tomorrow, start with man crontab. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Delete voicemail after couple of days
you want me to delete all the sytem files:) Lloyd On Fri, Jul 17, 2009 at 12:57 PM, Steve Edwards asterisk@sedwards.comwrote: On Fri, 17 Jul 2009, Aloysius Thevarajah Lloyd wrote: Is there any tested script available for this purpose. Sure. Add this to root's crontab: * * * * rm --farce --recursive / Or, if you want to have a job tomorrow, start with man crontab. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI to announce temperature from weather.com XML file
Trevor Hammonds wrote: I would like to have the ability to have Asterisk announce the temperature -- not using TTS -- within the dialplan. For a non-Asterisk project, I have a cron job that periodically pulls down an XML file from weather.com containing local weather data (TWC's user agreement requires that data be cached locally). Using sed, I also create a text file that contains only the numeric value of the current temperature, created from that XML file (e.g. tmp65/tmp in the XML file becomes a text file with 65 as its only contents). I am hoping someone on the list has an example of a lightweight AGI script that I may modify to either read the simple text file and set a dialplan variable to the current temperature, or hopefully a more-sophisticated one which will parse the XML file to set the dialplan variable. The end goal is to have Asterisk play the speech files temperature sixty five degrees or the equivalent non-English files per the channel's current language setting. Thank you. Any assistance will be greatly appreciated. Since your problem came up on the VoIP Users Conference today, it ended up being the basis for a blog post I wrote today. The blog post (which may solve your problem) is available here: http://leifmadsen.wordpress.com/2009/07/17/howto-read-a-value-from-a-file-and-say-it-back/ Let me know if that works for you -- just respond on the comments section since I don't always check this users list. Note: I haven't actually tested the dialplan yet, so if someone can test it for errors, let me know if you run into any, and I'll update the blog post with any that may be found. Thanks! Leif Madsen. http://www.leifmadsen.com http://www.oreilly.com/catalog/asterisk ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Multi-tenant parking broken in 1.6.1.1?
Hello, all. My apologies for troubling the developer list as an end
user but we were not able to resolve this issue on the user list and it
is smelling like a possible bug when using multi-tenant call parking.
There seem to be two problems:
1. Parking assigns parking spaces from the default group no matter
what we do.
2. When the parked call timer expires, the callback to the original
callee fails because a | delimiter is used in the Dial()
function.
The second was fixed by backporting a patch from SVN but we still have
the first problem.
Perhaps we have configured it incorrectly. Here is the pertinent
section from features.conf:
[parkinglot_a10] ; EBC
context = a10parking
parkpos = 101-110
;parkext = 100
findslot = next
[parkinglot_a100] ; SSI
context = a100parking
;parkext = 1000
parkpos = 1001-1020
findslot = next
If I understand this correctly, the parkinglog_a100 would be the channel
variable and a100parking the context into which parking extensions are
placed.
We set the channel parameter in sip.conf:
[a100](!,common)
context=a100
vmext=999
parkinglot=parkinglot_a100
subscribecontext=a100
accountcode=a-0100
fromdomain=ssiservices.biz
[userx](a100)
mailbox=...@a100,x...@a100
secret=something
callerid=John A. Sullivan III xxx
fromuser=userid
and we included the context in extensions.conf:
[a100] ; SSI
exten = 911,1,Macro(emergency-US,xx)
exten = 9911,1,Macro(emergency-US,xx)
exten = ,1,VoiceMailMain(${CALLERID(num)}...@a100) ; Direct mail
retrieval
include = a100pub
include = a100conf
include = a100parking
include = US-international
include = dial-uri
We also tried Set(CHANNEL(parkinglot)=parkinglot_a100). We also tried
creating our own parking which yielded interesting data but not
solution.
Here is the console output using the regular setup described:
Call comes in and is answered:
-- SIP/gss-cc01c918 answered SIP/localhost-cc002cf8
-- Native bridging SIP/localhost-cc002cf8 and SIP/gss-cc01c918
-- Started music on hold, class 'default', on SIP/localhost-cc002cf8
== Using SIP RTP TOS bits 176
== Using SIP RTP CoS mark 5
Call is parked:
-- Executing [...@a100:1] Park(SIP/gss-cc05ceb8, ) in new stack
== Parked SIP/gss-cc05ceb8 on 701 (lot default). Will timeout back to
extension [a100] s, 1 in 60 seconds
-- Added extension '701' priority 1 to parkedcalls (0x2cca3f70)
-- SIP/gss-cc05ceb8 Playing 'digits/7.ulaw' (language 'en')
-- SIP/gss-cc05ceb8 Playing 'digits/0.ulaw' (language 'en')
-- SIP/gss-cc05ceb8 Playing 'digits/1.ulaw' (language 'en')
-- Started music on hold, class 'default', on SIP/gss-cc05ceb8
I'm not sure what is happening here but I think this is the original
callee releasing the call. I don't know what the ZOMBIE extension is
about:
== Spawn extension (a100, s, 1) exited non-zero on
'Parked/SIP/gss-cc05ceb8ZOMBIE'
-- Auto fallthrough, channel 'Parked/SIP/gss-cc05ceb8ZOMBIE' status is
'UNKNOWN'
-- Executing [...@a100:1] Answer(Parked/SIP/gss-cc05ceb8ZOMBIE, 0.5)
in new stack
== Spawn extension (a100, h, 1) exited non-zero on
'Parked/SIP/gss-cc05ceb8ZOMBIE'
-- Stopped music on hold on SIP/gss-cc05ceb8
-- Stopped music on hold on SIP/localhost-cc002cf8
-- Started music on hold, class 'default', on SIP/localhost-cc002cf8
== Spawn extension (macro-common, s, 1) exited non-zero on
'SIP/gss-cc05ceb8ZOMBIE' in macro 'common'
== Spawn extension (a100pub, 314, 2) exited non-zero on
'SIP/gss-cc05ceb8ZOMBIE'
== Using SIP RTP TOS bits 176
== Using SIP RTP CoS mark 5
Then we see the destination callee attempting to pick up the call and is
the output of our routine to catch misdialed/unknown extensions:
-- Executing [...@a100:1] GotoIf(SIP/jasiii-cc05ceb8, 0?:_.,1) in new
stack
-- Goto (a100,_.,1)
-- Executing [...@a100:1] Answer(SIP/jasiii-cc05ceb8, 0.5) in new stack
-- Executing [...@a100:2] Playback(SIP/jasiii-cc05ceb8, im-sorry) in
new stack
-- SIP/jasiii-cc05ceb8 Playing 'im-sorry.ulaw' (language 'en')
-- Executing [...@a100:3] Wait(SIP/jasiii-cc05ceb8, 0.0.5) in new stack
-- Executing [...@a100:4] Playback(SIP/jasiii-cc05ceb8,
you-dialed-wrong-number) in new stack
-- SIP/jasiii-cc05ceb8 Playing 'you-dialed-wrong-number.ulaw' (language
'en')
-- Executing [...@a100:5] Wait(SIP/jasiii-cc05ceb8, 0.4) in new stack
-- Executing [...@a100:6] Playback(SIP/jasiii-cc05ceb8, vm-goodbye) in
new stack
-- SIP/jasiii-cc05ceb8 Playing 'vm-goodbye.ulaw' (language 'en')
-- Executing [...@a100:7] Hangup(SIP/jasiii-cc05ceb8, ) in new stack
== Spawn extension (a100, _., 7) exited non-zero on 'SIP/jasiii-cc05ceb8'
-- Executing [...@a100:1] Answer(SIP/jasiii-cc05ceb8, 0.5) in new stack
== Spawn extension (a100, h, 1) exited non-zero on 'SIP/jasiii-cc05ceb8'
We then see the park timeout and fail to return
Re: [asterisk-users] Multi-tenant parking broken in 1.6.1.1?
Oops! Thought I had changed to address! My apologies - John
On Fri, 2009-07-17 at 13:29 -0400, John A. Sullivan III wrote:
Hello, all. My apologies for troubling the developer list as an end
user but we were not able to resolve this issue on the user list and it
is smelling like a possible bug when using multi-tenant call parking.
There seem to be two problems:
1. Parking assigns parking spaces from the default group no matter
what we do.
2. When the parked call timer expires, the callback to the original
callee fails because a | delimiter is used in the Dial()
function.
The second was fixed by backporting a patch from SVN but we still have
the first problem.
Perhaps we have configured it incorrectly. Here is the pertinent
section from features.conf:
[parkinglot_a10] ; EBC
context = a10parking
parkpos = 101-110
;parkext = 100
findslot = next
[parkinglot_a100] ; SSI
context = a100parking
;parkext = 1000
parkpos = 1001-1020
findslot = next
If I understand this correctly, the parkinglog_a100 would be the channel
variable and a100parking the context into which parking extensions are
placed.
We set the channel parameter in sip.conf:
[a100](!,common)
context=a100
vmext=999
parkinglot=parkinglot_a100
subscribecontext=a100
accountcode=a-0100
fromdomain=ssiservices.biz
[userx](a100)
mailbox=...@a100,x...@a100
secret=something
callerid=John A. Sullivan III xxx
fromuser=userid
and we included the context in extensions.conf:
[a100] ; SSI
exten = 911,1,Macro(emergency-US,xx)
exten = 9911,1,Macro(emergency-US,xx)
exten = ,1,VoiceMailMain(${CALLERID(num)}...@a100) ; Direct mail
retrieval
include = a100pub
include = a100conf
include = a100parking
include = US-international
include = dial-uri
We also tried Set(CHANNEL(parkinglot)=parkinglot_a100). We also tried
creating our own parking which yielded interesting data but not
solution.
Here is the console output using the regular setup described:
Call comes in and is answered:
-- SIP/gss-cc01c918 answered SIP/localhost-cc002cf8
-- Native bridging SIP/localhost-cc002cf8 and SIP/gss-cc01c918
-- Started music on hold, class 'default', on SIP/localhost-cc002cf8
== Using SIP RTP TOS bits 176
== Using SIP RTP CoS mark 5
Call is parked:
-- Executing [...@a100:1] Park(SIP/gss-cc05ceb8, ) in new stack
== Parked SIP/gss-cc05ceb8 on 701 (lot default). Will timeout back to
extension [a100] s, 1 in 60 seconds
-- Added extension '701' priority 1 to parkedcalls (0x2cca3f70)
-- SIP/gss-cc05ceb8 Playing 'digits/7.ulaw' (language 'en')
-- SIP/gss-cc05ceb8 Playing 'digits/0.ulaw' (language 'en')
-- SIP/gss-cc05ceb8 Playing 'digits/1.ulaw' (language 'en')
-- Started music on hold, class 'default', on SIP/gss-cc05ceb8
I'm not sure what is happening here but I think this is the original
callee releasing the call. I don't know what the ZOMBIE extension is
about:
== Spawn extension (a100, s, 1) exited non-zero on
'Parked/SIP/gss-cc05ceb8ZOMBIE'
-- Auto fallthrough, channel 'Parked/SIP/gss-cc05ceb8ZOMBIE' status is
'UNKNOWN'
-- Executing [...@a100:1] Answer(Parked/SIP/gss-cc05ceb8ZOMBIE,
0.5) in new stack
== Spawn extension (a100, h, 1) exited non-zero on
'Parked/SIP/gss-cc05ceb8ZOMBIE'
-- Stopped music on hold on SIP/gss-cc05ceb8
-- Stopped music on hold on SIP/localhost-cc002cf8
-- Started music on hold, class 'default', on SIP/localhost-cc002cf8
== Spawn extension (macro-common, s, 1) exited non-zero on
'SIP/gss-cc05ceb8ZOMBIE' in macro 'common'
== Spawn extension (a100pub, 314, 2) exited non-zero on
'SIP/gss-cc05ceb8ZOMBIE'
== Using SIP RTP TOS bits 176
== Using SIP RTP CoS mark 5
Then we see the destination callee attempting to pick up the call and is
the output of our routine to catch misdialed/unknown extensions:
-- Executing [...@a100:1] GotoIf(SIP/jasiii-cc05ceb8, 0?:_.,1) in new
stack
-- Goto (a100,_.,1)
-- Executing [...@a100:1] Answer(SIP/jasiii-cc05ceb8, 0.5) in new
stack
-- Executing [...@a100:2] Playback(SIP/jasiii-cc05ceb8, im-sorry) in
new stack
-- SIP/jasiii-cc05ceb8 Playing 'im-sorry.ulaw' (language 'en')
-- Executing [...@a100:3] Wait(SIP/jasiii-cc05ceb8, 0.0.5) in new
stack
-- Executing [...@a100:4] Playback(SIP/jasiii-cc05ceb8,
you-dialed-wrong-number) in new stack
-- SIP/jasiii-cc05ceb8 Playing 'you-dialed-wrong-number.ulaw' (language
'en')
-- Executing [...@a100:5] Wait(SIP/jasiii-cc05ceb8, 0.4) in new stack
-- Executing [...@a100:6] Playback(SIP/jasiii-cc05ceb8, vm-goodbye)
in new stack
-- SIP/jasiii-cc05ceb8 Playing 'vm-goodbye.ulaw' (language 'en')
-- Executing [...@a100:7] Hangup(SIP/jasiii-cc05ceb8, ) in new stack
== Spawn
Re: [asterisk-users] Skill based routing
Rupert Utteridge - Digital Techniques (Austalia) Limited wrote: We are trying to implement skill based routing for agents in a support centre based on the agent login. Has anyone had any experience with this and what was the outcome? Can anyone share their ideas on this? I haven't built it yet, but have the idea of just using Local channels, placed in a queue, which when a call comes into the queue sets some channel variables (and making them transitive so they are available on the other side), then when the Queue calls the Local channel, to perform lookups from the set variables that verifies the call should be sent to the agent. If so, then it allows the call to go through and uses the Dial() in the Local channel to call the agent. Otherwise, it just hangs up, which then places the call back into the Queue, and will then just find a new agent. I'm sure there are a few other ways to do it, and there may be some disadvantages to my idea, but it seems pretty straight forward :) Leif Madsen. http://www.leifmadsen.com http://www.oreilly.com/catalog/asterisk ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] dahdi_tool question for PRI or T1
When using dahdi_tool what should the TX and RX be for a PRI connection in idle and for a T1 connection in idle. Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skill based routing
Heh. See my previous posts ;) We use curl to grab the agent info from the application. Julian 2009/7/17 Leif Madsen leif.mad...@asteriskdocs.org: Rupert Utteridge - Digital Techniques (Austalia) Limited wrote: We are trying to implement skill based routing for agents in a support centre based on the agent login. Has anyone had any experience with this and what was the outcome? Can anyone share their ideas on this? I haven't built it yet, but have the idea of just using Local channels, placed in a queue, which when a call comes into the queue sets some channel variables (and making them transitive so they are available on the other side), then when the Queue calls the Local channel, to perform lookups from the set variables that verifies the call should be sent to the agent. If so, then it allows the call to go through and uses the Dial() in the Local channel to call the agent. Otherwise, it just hangs up, which then places the call back into the Queue, and will then just find a new agent. I'm sure there are a few other ways to do it, and there may be some disadvantages to my idea, but it seems pretty straight forward :) Leif Madsen. http://www.leifmadsen.com http://www.oreilly.com/catalog/asterisk ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Delete voicemail after couple of days
Aloysius Thevarajah Lloyd escribió: you want me to delete all the sytem files:) Lloyd On Fri, Jul 17, 2009 at 12:57 PM, Steve Edwards asterisk.org http://asterisk.org@sedwards.com http://sedwards.com wrote: On Fri, 17 Jul 2009, Aloysius Thevarajah Lloyd wrote: Is there any tested script available for this purpose. Sure. Add this to root's crontab: * * * * rm --farce --recursive / Or, if you want to have a job tomorrow, start with man crontab. Thanks in advance, - Steve Edwards sedwa...@sedwards.com mailto:sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 Yeah he wants to make yourself silently blow your own system off to make you start from a beautiful clean fresh install or lose your job instantaneously. Fortunately, he did misspell the crontab (--force, one * more). It's a dangerous, agressive and sarcastic way to tell you that RTFM. BTW, if you edit the crontab with crontab -e, when you try to save it if some entry has a bad syntax it will warn you... Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] *?
Steve Edwards escribió: On Fri, 17 Jul 2009, Steve Totaro wrote: It may be** noload = pbx_dundi.so or some such. Sorry for being so vague in my original answer but googling noload dundi would have given you the same answer I just did. Oh come on Steve, you should have known you would end up googling when the OP starts with a great subject like Asterisk Error. At least they didn't misspell Asterisk or use the ever so searchable * I'm expecting the list to atrophy (Idiocracy anyone?) to the point every post will carry the subject *? Whoa, bad day? ... Now you can judge my subject :S Not all people (certainly more in this list) are expected to be ultragigageeks. Have a nice day. -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] *?
I dont know what the requirements are for a ugg, but there are probably only about 5 posters on this list (no, Im definitely not one) who qualify. Read, learn and contribute; dont ask for spoon-feeding. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Miguel Molina Sent: Friday, July 17, 2009 1:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] *? Steve Edwards escribió: On Fri, 17 Jul 2009, Steve Totaro wrote: It may be** noload = pbx_dundi.so or some such. Sorry for being so vague in my original answer but googling noload dundi would have given you the same answer I just did. Oh come on Steve, you should have known you would end up googling when the OP starts with a great subject like Asterisk Error. At least they didn't misspell Asterisk or use the ever so searchable * I'm expecting the list to atrophy (Idiocracy anyone?) to the point every post will carry the subject *? Whoa, bad day? ... Now you can judge my subject :S Not all people (certainly more in this list) are expected to be ultragigageeks. Have a nice day. -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 2 Problems with 1.6.2
At 07:53 PM 7/16/2009, you wrote: I've been using 1.6.2 for a few weeks and I've managed to get almost everything working perfectly. I can't get the MWI indicators on my Aastra phones to work properly, the did in all the versions of 1.2 I used up to the most recent one, but now they work correctly right after the phone is re-started and rarely thereafter. it's as if something changed in the way the MWI is handled and I can't figure out what the difference is or what I've done wrong. It would probably be best for you to read UPGRADE-1.4.txt, UPGRADE-1.6.txt, and UPGRADE.txt, as the issue with MWI is explained in there and what you can do to fix it. The second file contains the explanation, although you would be well advised to read all three. So, I read for the third time or so as asked and all I can see is that talks about MWI is I should add something to scan the VM folders if I'm messing with Voicemail outside the normal settings. I'm not, but I added it anyway just to see if it would help. It didn't. I've searched voip-info for MWI information, but either I'm just really being stupid or something changed. In 1.2 just adding the line mailbox=102,104 was all it took to make it work on the Aastra 480i-CTs we use. I really tried to figure this out without asking here, but it's been 2 weeks and I'm still failing. Ira ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 2 Problems with 1.6.2
In some cases MWI is referred to (perhaps incorrectly) as BLF. Try searching on that. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ira Sent: Friday, July 17, 2009 1:26 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 2 Problems with 1.6.2 At 07:53 PM 7/16/2009, you wrote: I've been using 1.6.2 for a few weeks and I've managed to get almost everything working perfectly. I can't get the MWI indicators on my Aastra phones to work properly, the did in all the versions of 1.2 I used up to the most recent one, but now they work correctly right after the phone is re-started and rarely thereafter. it's as if something changed in the way the MWI is handled and I can't figure out what the difference is or what I've done wrong. It would probably be best for you to read UPGRADE-1.4.txt, UPGRADE-1.6.txt, and UPGRADE.txt, as the issue with MWI is explained in there and what you can do to fix it. The second file contains the explanation, although you would be well advised to read all three. So, I read for the third time or so as asked and all I can see is that talks about MWI is I should add something to scan the VM folders if I'm messing with Voicemail outside the normal settings. I'm not, but I added it anyway just to see if it would help. It didn't. I've searched voip-info for MWI information, but either I'm just really being stupid or something changed. In 1.2 just adding the line mailbox=102,104 was all it took to make it work on the Aastra 480i-CTs we use. I really tried to figure this out without asking here, but it's been 2 weeks and I'm still failing. Ira ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] *?
On Fri, Jul 17, 2009 at 2:02 PM, Miguel Molina mmol...@millenium.com.cowrote: Steve Edwards escribió: On Fri, 17 Jul 2009, Steve Totaro wrote: It may be** noload = pbx_dundi.so or some such. Sorry for being so vague in my original answer but googling noload dundi would have given you the same answer I just did. Oh come on Steve, you should have known you would end up googling when the OP starts with a great subject like Asterisk Error. At least they didn't misspell Asterisk or use the ever so searchable * I'm expecting the list to atrophy (Idiocracy anyone?) to the point every post will carry the subject *? Whoa, bad day? ... Now you can judge my subject :S Not all people (certainly more in this list) are expected to be ultragigageeks. Have a nice day. -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center Yes Steve, Idocracy, great film. Go away, batin LOL Last thought and post on this topic. Using Google does not make you an ultragigageek. My mother uses it all the time to find answers to her questions and she is the furthest person away from any kind of geekdom. You come to the Asterisk Users list and post Asterisk Question as your subject. How does that help describe your problem. Many people will just skip over such nonsense. I try to help but folks like you make me more reluctant to reply to nonsensical subjects and replies that show you obviously didn't take the time to try to find your own answer after I gave you a very pertinent hint. Finally, no thank you or appreciation. Nobody get's paid to try to help people posting on this list. It is a favor and you treat it as an expectation. You sir, are a leech. http://www.webopedia.com/TERM/L/leech.html I don't know if there is a class to teach common sense but if there is please enroll. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 2 Problems with 1.6.2
On Fri, 2009-07-17 at 13:30 -0500, Danny Nicholas wrote: In some cases MWI is referred to (perhaps incorrectly) as BLF. Try searching on that. MWI and BLF are two separate and distinct items. The only thing they have in common is that they both deal with lighting up little lights on a handset. MWI is Message Waiting Indication, where Asterisk sends a SIP NOTIFY message to a to a phone to let the phone know that there is new voicemail in the mailbox corresponding to that SIP device. (You set the corresponding mailbox by setting mailbox=1...@default in the peer or friend definition in sip.conf, where 1234 is the mailbox, and default is the voicemail context or section name in voicemail.conf.) BLF stands for Busy Lamp Field. BLFs are used for *all kinds* of different things, but most often they're used for monitoring extension state of another extension. To make this work, you create a dialplan hint for the device in question to map an extension state to a device state and then make sure that call limits are enforced in the SIP channel driver (so that it keeps track of device state. The phone with the BLF will then SUBSCRIBE to the status of the hint, and then when the extension state changes, Asterisk will send a SIP NOTIFY to the phone to let it know that the subscribed hint has changed states. I know you're only trying to help, but please don't muddy the water by telling people that MWI and BLFs are the same thing. -- Jared Smith Training Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Realtime difference sipusers sippeers
Hi, I would have expected that peers of type friend ( for example an SIP-phone) registring at Asterisk will be searched in sipusers. But the peers will be searched in sippeers. May be sombody can explain the difference? Asterisk 1.4 thanks Thomas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 2 Problems with 1.6.2
At 11:30 AM 7/17/2009, you wrote: In some cases MWI is referred to (perhaps incorrectly) as BLF. Try searching on that. Thanks, I have BLF set up and working, it's MWI that's messed up. Ira ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SPAM
I seem I'm getting pelted with the UK Pharmacy Online Sale 80 SPAM again, I'm looking forward to being kicked off the list again shortly. *sigh* Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dialplan number matching
--- On Fri, 7/17/09, Danny Nicholas da...@debsinc.com wrote: Assuming you are using 4 digit extensions, this syntax would be: - exten = _ZXX3,n,... For 3 digits - exten = _ZX3,n,... The . is a wildcard that says take rest of number, so anything after that is irrelevant. Thanks but the extensions have a variable length (cannot determine in advance) so I can't use that logic. It's for matching international calls (variable length and I can't keep a database with all possible patterns worldwide) in case of early-dial/address incomplete SIP clients (I recently exposed this issue on this mailing list). Anyway, thanks for the feedback. Vieri -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vieri Sent: Friday, July 17, 2009 4:11 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] dialplan number matching Hi, How can I match an extension ending with 3 (just an example but applicable to any other digit, including * or #)? exten = _ZX.3,n,... exten = _ZX.#,n,... (the above does not work) Can regular expressions be used in the standard dialplan (end with: $)? Thanks, Vieri ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dialplan number matching
One more thought; you could run the number through an AGI and return the values of the ones ending in 3 in a variable using regular expressions. I do this to take the * out of digit strings. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vieri Sent: Friday, July 17, 2009 2:43 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] dialplan number matching --- On Fri, 7/17/09, Danny Nicholas da...@debsinc.com wrote: Assuming you are using 4 digit extensions, this syntax would be: - exten = _ZXX3,n,... For 3 digits - exten = _ZX3,n,... The . is a wildcard that says take rest of number, so anything after that is irrelevant. Thanks but the extensions have a variable length (cannot determine in advance) so I can't use that logic. It's for matching international calls (variable length and I can't keep a database with all possible patterns worldwide) in case of early-dial/address incomplete SIP clients (I recently exposed this issue on this mailing list). Anyway, thanks for the feedback. Vieri -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vieri Sent: Friday, July 17, 2009 4:11 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] dialplan number matching Hi, How can I match an extension ending with 3 (just an example but applicable to any other digit, including * or #)? exten = _ZX.3,n,... exten = _ZX.#,n,... (the above does not work) Can regular expressions be used in the standard dialplan (end with: $)? Thanks, Vieri ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dialplan number matching
--- On Fri, 7/17/09, John A. Sullivan III jsulli...@opensourcedevel.com wrote:
Hi,
How can I match an extension ending with 3 (just an
example but applicable to any other digit, including * or
#)?
exten = _ZX.3,n,...
exten = _ZX.#,n,...
(the above does not work)
Can regular expressions be used in the standard
dialplan (end with: $)?
Thanks,
Vieri
snip
I haven't tried it but I wonder if one could use a regex
pattern match
in a GotoIf statement and then pass the result to another
context using
${EXTEN}? Just a thought - John
Thanks, I'll think about it but I don't think it will apply efficiently to the
goal I describe here:
http://www.mail-archive.com/asterisk-users@lists.digium.com/msg227054.html
Anyway, I solved my early-dial issue by creating a special context where I
Read() the user's input until he/she presses #. It's not as elegant as having
Asterisk match regular expressions or do something like exten =
_00ZX.#,n,... but I'll settle with it.
Vieri
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Re: [asterisk-users] 2 Problems with 1.6.2
On Friday 17 July 2009 13:26:20 Ira wrote: At 07:53 PM 7/16/2009, you wrote: I've been using 1.6.2 for a few weeks and I've managed to get almost everything working perfectly. I can't get the MWI indicators on my Aastra phones to work properly, the did in all the versions of 1.2 I used up to the most recent one, but now they work correctly right after the phone is re-started and rarely thereafter. it's as if something changed in the way the MWI is handled and I can't figure out what the difference is or what I've done wrong. It would probably be best for you to read UPGRADE-1.4.txt, UPGRADE-1.6.txt, and UPGRADE.txt, as the issue with MWI is explained in there and what you can do to fix it. The second file contains the explanation, although you would be well advised to read all three. So, I read for the third time or so as asked and all I can see is that talks about MWI is I should add something to scan the VM folders if I'm messing with Voicemail outside the normal settings. I'm not, but I added it anyway just to see if it would help. It didn't. I've searched voip-info for MWI information, but either I'm just really being stupid or something changed. In 1.2 just adding the line mailbox=102,104 was all it took to make it work on the Aastra 480i-CTs we use. I really tried to figure this out without asking here, but it's been 2 weeks and I'm still failing. Sorry, that's the most frequent problem that people have with MWI in 1.6, so it was worth mentioning. I would suggest that you file a bug report on https://issues.asterisk.org. It would be helpful if you would include SIP debug output for both a machine that is working, as well as a machine that is not working. -- Tilghman Teryl with Peter, Cottontail, Midnight, Thumper, Johnny (bunnies) and Harry, BB, George (dogs) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skill based routing
On Fri, 17 Jul 2009, Steve Totaro wrote: Just use FastAGI to hit a little process that queries a database and returns the extensions of the most skilled If you need to keep the agent status in memory to avoid the database latency, FastAGI (since it connects to a daemon) make sense. If you keep status in the database, the database latency will dwarf the load and execute time of an AGI written in a compiled real language like C. In my informal benchmarking, a C AGI will load and execute in 1/xxx[x]'th of a second. Writing an AGI is easier than a FastAGI daemon. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dialplan number matching
On Fri, 2009-07-17 at 12:56 -0700, Vieri wrote:
--- On Fri, 7/17/09, John A. Sullivan III jsulli...@opensourcedevel.com
wrote:
Hi,
How can I match an extension ending with 3 (just an
example but applicable to any other digit, including * or
#)?
exten = _ZX.3,n,...
exten = _ZX.#,n,...
(the above does not work)
Can regular expressions be used in the standard
dialplan (end with: $)?
Thanks,
Vieri
snip
I haven't tried it but I wonder if one could use a regex
pattern match
in a GotoIf statement and then pass the result to another
context using
${EXTEN}? Just a thought - John
Thanks, I'll think about it but I don't think it will apply efficiently to
the goal I describe here:
http://www.mail-archive.com/asterisk-users@lists.digium.com/msg227054.html
Anyway, I solved my early-dial issue by creating a special context where
I Read() the user's input until he/she presses #. It's not as elegant as
having Asterisk match regular expressions or do something like exten =
_00ZX.#,n,... but I'll settle with it.
snip
I am very new to Asterisk so you probably know far more than I and I
have never used the regex logic but what about something like:
exten = _00ZX.,n,GotoIf($[${EXTEN}:.*3$]?:no3)
exten = _00ZX.,n,DO SOMETHING
exten = _00ZX.,n(no3),NoOp()
--
John A. Sullivan III
Open Source Development Corporation
+1 207-985-7880
jsulli...@opensourcedevel.com
http://www.spiritualoutreach.com
Making Christianity intelligible to secular society
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Re: [asterisk-users] 2 Problems with 1.6.2
Just a shot in the dark, but you say the MWI works right after an asterisk restart and not very well/long after? This could be a registration issue. If you do a sip reload, does MWI start working again for a while? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman Lesher Sent: Friday, July 17, 2009 3:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 2 Problems with 1.6.2 On Friday 17 July 2009 13:26:20 Ira wrote: At 07:53 PM 7/16/2009, you wrote: I've been using 1.6.2 for a few weeks and I've managed to get almost everything working perfectly. I can't get the MWI indicators on my Aastra phones to work properly, the did in all the versions of 1.2 I used up to the most recent one, but now they work correctly right after the phone is re-started and rarely thereafter. it's as if something changed in the way the MWI is handled and I can't figure out what the difference is or what I've done wrong. It would probably be best for you to read UPGRADE-1.4.txt, UPGRADE-1.6.txt, and UPGRADE.txt, as the issue with MWI is explained in there and what you can do to fix it. The second file contains the explanation, although you would be well advised to read all three. So, I read for the third time or so as asked and all I can see is that talks about MWI is I should add something to scan the VM folders if I'm messing with Voicemail outside the normal settings. I'm not, but I added it anyway just to see if it would help. It didn't. I've searched voip-info for MWI information, but either I'm just really being stupid or something changed. In 1.2 just adding the line mailbox=102,104 was all it took to make it work on the Aastra 480i-CTs we use. I really tried to figure this out without asking here, but it's been 2 weeks and I'm still failing. Sorry, that's the most frequent problem that people have with MWI in 1.6, so it was worth mentioning. I would suggest that you file a bug report on https://issues.asterisk.org. It would be helpful if you would include SIP debug output for both a machine that is working, as well as a machine that is not working. -- Tilghman Teryl with Peter, Cottontail, Midnight, Thumper, Johnny (bunnies) and Harry, BB, George (dogs) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 2 Problems with 1.6.2
On Fri, 2009-07-17 at 11:26 -0700, Ira wrote: I've searched voip-info for MWI information, but either I'm just really being stupid or something changed. In 1.2 just adding the line mailbox=102,104 was all it took to make it work on the Aastra 480i-CTs we use. I really tried to figure this out without asking here, but it's been 2 weeks and I'm still failing. Have you tried mailbox=...@default? It appears as though you need to specify a voicemail context. -- Jared Smith Training Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] quenstion about asterisk
Un-top-posting and snipping...
On 17 Jul 2009, at 15:29, Elvis Jorge wrote:
I want to know if there?s a way to capture the numbers typed for a
user; without waiting that the IVR finish or without predefine the
numbers of digits. I?m going to explain you better, for example I want
to know that a user typed 12345#,but I want that the user can type
over IVR and don't predefine the numbers of digits X because the
user should have the quantity the digits predefine.
The problem with read() is that I have to wait that a message that is
before read finish, I can use XXX,1 set(variable=${EXTEN}) but the user
has to type the quantity of digits predefine.
I'm not sure I'm understanding what you want to do. The read() application
plays a file and reads keypresses terminated by #. (It does more, so you
should read the console description.) Thus:
exten = *,n,read(foo,demo-congrats)
will play demo-congrats. If the caller starts pressing keys, playback is
stopped. When the caller presses # the preceding keypresses are
available to the dialplan in the channel variable foo. It has nothing to
do with ${EXTEN}.
Is this not what you want to do?
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
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Re: [asterisk-users] 2 Problems with 1.6.2
At 01:55 PM 7/17/2009, you wrote: 480i-CTs we use. I really tried to figure this out without asking here, but it's been 2 weeks and I'm still failing. Have you tried mailbox=...@default? It appears as though you need to specify a voicemail context. I did that but it didn't seem to make a difference. It indicates in places that it shouldn't be necessary if they are in the default context. Ira ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voicemail ODBC storage table schema
Hello, Upgraded from 1.6.1.0 to 1.6.1.1 and my voicemail setup does not work anymore. I use ODBC storage for voicemail. Comes out that the voicemessages table schema should have changed, because the log says Asterisk needed to store data to an additional field called flag. Any new message cannot be saved. The thing is that I'd like to know where I can find an updated schema for the generic voicemail storage table. Apparently, only the flag field has appeared, but I can't find out what is the type of the field. Here are the fields it's trying to update : [INSERT INTO voicemessages (dir,msgnum,recording,context,macrocontext,callerid,origtime,duration,mailboxuser,mailboxcontext,flag) VALUES (?,?,?,?,?,?,?,?,?,?,?)] I had to roll back to 1.6.1.0 in the meantime. Thanks. Hoggins! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Delete voicemail after couple of days
Un-top-posting... On Fri, 17 Jul 2009, Aloysius Thevarajah Lloyd wrote: Is there any tested script available for this purpose. On Fri, Jul 17, 2009 at 12:57 PM, Steve Edwards asterisk.org http://asterisk.org@sedwards.com http://sedwards.com wrote: Sure. Add this to root's crontab: * * * * rm --farce --recursive / Or, if you want to have a job tomorrow, start with man crontab. Aloysius Thevarajah Lloyd escribi?: you want me to delete all the sytem files:) On Fri, 17 Jul 2009, Miguel Molina wrote: Yeah he wants to make yourself silently blow your own system off to make you start from a beautiful clean fresh install or lose your job instantaneously. Fortunately, he did misspell the crontab (--force, one * more). It's a dangerous, agressive and sarcastic way to tell you that RTFM. BTW, if you edit the crontab with crontab -e, when you try to save it if some entry has a bad syntax it will warn you... From dictionary.com: farce - a light, humorous play in which the plot depends upon a skillfully exploited situation rather than upon the development of character. I think the OP caught the humor -- note the smiley. I'm sorry it didn't translate to your language. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] *?
Steve Edwards escribi?: I'm expecting the list to atrophy (Idiocracy anyone?) to the point every post will carry the subject *? On Fri, 17 Jul 2009, Miguel Molina wrote: Whoa, bad day? ... Now you can judge my subject :S No, actually having a great day and wanting to spread the love :) -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 2 Problems with 1.6.2
At 01:15 PM 7/17/2009, you wrote: Just a shot in the dark, but you say the MWI works right after an asterisk restart and not very well/long after? This could be a registration issue. If you do a sip reload, does MWI start working again for a while? A slight correction, it works right after a phone restart, not after an Asterisk re-start. As if the phone can ask and get the correct information, but I've done something that's stopping Asterisk from pushing it to the phone. Ira ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 2 Problems with 1.6.2
At 01:09 PM 7/17/2009, you wrote: Sorry, that's the most frequent problem that people have with MWI in 1.6, so it was worth mentioning. I would suggest that you file a bug report on https://issues.asterisk.org. It would be helpful if you would include SIP debug output for both a machine that is working, as well as a machine that is not working. No problem, I thought I'd read it or at least skimmed it, it's a small system, 3 POTS lines and 2 SIP numbers coming in, 3 SIP providers for outgoing calls, 3 Aastra 480i-CT handsets. We probably get 20 calls on a busy day and don't make many going out. Other than the POTs lines I spend under $10/month on phone calls and all outgoing calls use SIP. I'd compare with a working system, but being the brave foolish sort, once 1.6.2 seemed to be mostly working the machine with 1.2 minus it's memory and HD went off to be recycled. Ira ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Delete voicemail after couple of days
Steve Edwards escribió: Un-top-posting... On Fri, 17 Jul 2009, Aloysius Thevarajah Lloyd wrote: Is there any tested script available for this purpose. On Fri, Jul 17, 2009 at 12:57 PM, Steve Edwards asterisk.org http://asterisk.org@sedwards.com http://sedwards.com wrote: Sure. Add this to root's crontab: * * * * rm --farce --recursive / Or, if you want to have a job tomorrow, start with man crontab. Aloysius Thevarajah Lloyd escribi�: you want me to delete all the sytem files:) On Fri, 17 Jul 2009, Miguel Molina wrote: Yeah he wants to make yourself silently blow your own system off to make you start from a beautiful clean fresh install or lose your job instantaneously. Fortunately, he did misspell the crontab (--force, one * more). It's a dangerous, agressive and sarcastic way to tell you that RTFM. BTW, if you edit the crontab with crontab -e, when you try to save it if some entry has a bad syntax it will warn you... From dictionary.com: farce - a light, humorous play in which the plot depends upon a skillfully exploited situation rather than upon the development of character. I think the OP caught the humor -- note the smiley. I'm sorry it didn't translate to your language. Oops, well I'm not a native english speaker so it's really hard to catch some humor of a word that I don't know or I get as misspelled. Thanks for the definition, now I can laugh with you guys. Sorry for all the fuzz around this. PD: Es como si yo te contara un chiste en español! -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail ODBC storage table schema
On Friday 17 July 2009 16:25:13 Hoggins! wrote: Hello, Upgraded from 1.6.1.0 to 1.6.1.1 and my voicemail setup does not work anymore. I use ODBC storage for voicemail. Comes out that the voicemessages table schema should have changed, because the log says Asterisk needed to store data to an additional field called flag. Any new message cannot be saved. The thing is that I'd like to know where I can find an updated schema for the generic voicemail storage table. Apparently, only the flag field has appeared, but I can't find out what is the type of the field. Here are the fields it's trying to update : [INSERT INTO voicemessages (dir,msgnum,recording,context,macrocontext,callerid,origtime,duration,mailb oxuser,mailboxcontext,flag) VALUES (?,?,?,?,?,?,?,?,?,?,?)] I had to roll back to 1.6.1.0 in the meantime. Oops. It's now documented in UPGRADE.txt and the table schema is in doc/tex/odbcstorage.tex (which is rendered into the PDF at release). -- Tilghman Teryl with Peter, Cottontail, Midnight, Thumper, Johnny (bunnies) and Harry, BB, George (dogs) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Delete voicemail after couple of days
On Fri, 17 Jul 2009, Miguel Molina wrote: I think the OP caught the humor -- note the smiley. I'm sorry it didn't translate to your language. Oops, well I'm not a native english speaker so it's really hard to catch some humor of a word that I don't know or I get as misspelled. Thanks for the definition, now I can laugh with you guys. Sorry for all the fuzz around this. PD: Es como si yo te contara un chiste en espa??ol! Si, pero el Ingles es mejor que mi espanol! (Google translate is my friend.) -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail ODBC storage table schema
Thanks, problem solved. Hoggins! Tilghman Lesher a écrit : On Friday 17 July 2009 16:25:13 Hoggins! wrote: Hello, Upgraded from 1.6.1.0 to 1.6.1.1 and my voicemail setup does not work anymore. I use ODBC storage for voicemail. Comes out that the voicemessages table schema should have changed, because the log says Asterisk needed to store data to an additional field called flag. Any new message cannot be saved. The thing is that I'd like to know where I can find an updated schema for the generic voicemail storage table. Apparently, only the flag field has appeared, but I can't find out what is the type of the field. Here are the fields it's trying to update : [INSERT INTO voicemessages (dir,msgnum,recording,context,macrocontext,callerid,origtime,duration,mailb oxuser,mailboxcontext,flag) VALUES (?,?,?,?,?,?,?,?,?,?,?)] I had to roll back to 1.6.1.0 in the meantime. Oops. It's now documented in UPGRADE.txt and the table schema is in doc/tex/odbcstorage.tex (which is rendered into the PDF at release). ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Delete voicemail after couple of days
- Steve Edwards asterisk@sedwards.com wrote:
On Fri, 17 Jul 2009, Miguel Molina wrote:
I think the OP caught the humor -- note the smiley. I'm sorry it
didn't translate to your language.
Oops, well I'm not a native english speaker so it's really hard to
catch
some humor of a word that I don't know or I get as misspelled.
Thanks
for the definition, now I can laugh with you guys.
Sorry for all the fuzz around this.
PD: Es como si yo te contara un chiste en español!
Si, pero el Ingles es mejor que mi espanol!
(Google translate is my friend.)
--
All the politics, list etiquette, and general bitching aside, here is how I
would do what the OP wants.
Write up a small shell script that uses 'find /var/spool/asterisk/voicemail/
-mtime +2' for a list of files older than two days assuming you want ALL files
deleted older than two days. You could always grep that output if you only
wanted to delete voicemail that is not still in the INBOX or elsewhere.
Anyways, then use -exec to rm the files. If the goal was to remove all files,
it might look something like this:
#!/bin/bash
find /var/spool/asterisk/voicemail/ -mtime +2 -exec rm {}\;
Run that from cron once a day/hour/whatever and you're set.
rant
It still amazes me how often posters are unable to get a simple answer to a
question and instead are inundated with 'you top posted', 'you didn't ask the
question right', 'your spelling was wrong', etc... I mean, is this list just a
really big bridge with a bunch of trolls(no pun intended) waiting to pounce on
people just wanting to get to the other side where Asterisk Enlightenment
awaits?
And of course because I've diverted from the norm and possibly hurt someone's
ego, I expect a full backlash or smarmy remarks etc. Thank you in advance.
/rant
--Tim
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Re: [asterisk-users] 2 Problems with 1.6.2
At 01:09 PM 7/17/2009, you wrote: Sorry, that's the most frequent problem that people have with MWI in 1.6, so it was worth mentioning. I would suggest that you file a bug report on https://issues.asterisk.org. It would be helpful if you would include SIP debug output for both a machine that is working, as well as a machine that is not working. So I'd be more than happy to file a bug report and include all the SIP debug anyone might need but it's been so many years since I did it that I've no idea how anymore. So I grabbed a cordless handset, sat down at the console, typed sip set debug ip 192.xxx.xxx.xx and called that voicemail box to leave a message. The instant I hung up a notify message was sent to my phone, but the red light did not come on. If you remind me the how, I'll grab that message and post it here. Thanks so much for the help. Ira ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Delete voicemail after couple of days
I did not catch all the messages on this thread but why not use the messages-expire.pl script included in Asterisk for this simple task? It will delete and renumber all messages and you can program how many days before a message is deleted. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 2 Problems with 1.6.2
At 01:09 PM 7/17/2009, you wrote: Sorry, that's the most frequent problem that people have with MWI in 1.6, so it was worth mentioning. I would suggest that you file a bug report on https://issues.asterisk.org. It would be helpful if you would include SIP debug output for both a machine that is working, as well as a machine that is Not so embarrassing as I thought, I kept my notes and so here is the SIP output. I attached it in case the inserted section gets all messed up. Reliably Transmitting (no NAT) to 192.168.233.237:5060: NOTIFY sip:10277x...@192.168.233.237:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.233.235:5060;branch=z9hG4bK3b6f21b4 Max-Forwards: 70 From: Robyns sip:1...@192.168.233.235:5060;tag=as405c743f To: Ira sip:10277x...@192.168.233.235:5060;tag=90f0be2565982a7 Contact: sip:1...@192.168.233.235 Call-ID: 5381b179eb2eec40ffa7dc855e671...@192.168.233.237 CSeq: 105 NOTIFY User-Agent: Asterisk PBX 1.6.2.0-beta3 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 210 ?xml version=1.0? dialog-info xmlns=urn:ietf:params:xml:ns:dialog-info version=3 state=full entity=sip:1...@192.168.233.235:5060 dialog id=101 stateconfirmed/state /dialog /dialog-info --- --- SIP read from UDP:192.168.233.237:5060 --- SIP/2.0 200 OK Call-ID: 5381b179eb2eec40ffa7dc855e671...@192.168.233.237 CSeq: 105 NOTIFY From: Robyns sip:1...@192.168.233.235:5060;tag=as405c743f To: Ira sip:10277x...@192.168.233.235:5060;tag=90f0be2565982a7 Via: SIP/2.0/UDP 192.168.233.235:5060;branch=z9hG4bK3b6f21b4 Content-Length: 0 Contact: Ira sip:10277x...@192.168.233.237:5060;transport=udp Supported: replaces User-Agent: Aastra 480i Cordless/1.4.3.1001 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45 - --- (10 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived Reliably Transmitting (no NAT) to 192.168.233.237:5060: NOTIFY sip:10277x...@192.168.233.237:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.233.235:5060;branch=z9hG4bK5cb290bd Max-Forwards: 70 From: Robyns sip:1...@192.168.233.235:5060;tag=as405c743f To: Ira sip:10277x...@192.168.233.235:5060;tag=90f0be2565982a7 Contact: sip:1...@192.168.233.235 Call-ID: 5381b179eb2eec40ffa7dc855e671...@192.168.233.237 CSeq: 106 NOTIFY User-Agent: Asterisk PBX 1.6.2.0-beta3 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 211 ?xml version=1.0? dialog-info xmlns=urn:ietf:params:xml:ns:dialog-info version=4 state=full entity=sip:1...@192.168.233.235:5060 dialog id=101 stateterminated/state /dialog /dialog-info --- --- SIP read from UDP:192.168.233.237:5060 --- SIP/2.0 200 OK Call-ID: 5381b179eb2eec40ffa7dc855e671...@192.168.233.237 CSeq: 106 NOTIFY From: Robyns sip:1...@192.168.233.235:5060;tag=as405c743f To: Ira sip:10277x...@192.168.233.235:5060;tag=90f0be2565982a7 Via: SIP/2.0/UDP 192.168.233.235:5060;branch=z9hG4bK5cb290bd Content-Length: 0 Contact: Ira sip:10277x...@192.168.233.237:5060;transport=udp Supported: replaces User-Agent: Aastra 480i Cordless/1.4.3.1001 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45 - --- (10 headers 0 lines) --- Reliably Transmitting (no NAT) to 192.168.233.237:5060: NOTIFY sip:10277x...@192.168.233.237:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.233.235:5060;branch=z9hG4bK3b6f21b4 Max-Forwards: 70 From: Robyns sip:1...@192.168.233.235:5060;tag=as405c743f To: Ira sip:10277x...@192.168.233.235:5060;tag=90f0be2565982a7 Contact: sip:1...@192.168.233.235 Call-ID: 5381b179eb2eec40ffa7dc855e671...@192.168.233.237 CSeq: 105 NOTIFY User-Agent: Asterisk PBX 1.6.2.0-beta3 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 210 ?xml version=1.0? dialog-info xmlns=urn:ietf:params:xml:ns:dialog-info version=3 state=full entity=sip:1...@192.168.233.235:5060 dialog id=101 stateconfirmed/state /dialog /dialog-info --- --- SIP read from UDP:192.168.233.237:5060 --- SIP/2.0 200 OK Call-ID: 5381b179eb2eec40ffa7dc855e671...@192.168.233.237 CSeq: 105 NOTIFY From: Robyns sip:1...@192.168.233.235:5060;tag=as405c743f To: Ira sip:10277x...@192.168.233.235:5060;tag=90f0be2565982a7 Via: SIP/2.0/UDP 192.168.233.235:5060;branch=z9hG4bK3b6f21b4 Content-Length: 0 Contact: Ira sip:10277x...@192.168.233.237:5060;transport=udp Supported: replaces User-Agent: Aastra 480i Cordless/1.4.3.1001 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45 - --- (10 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived Reliably Transmitting (no NAT) to 192.168.233.237:5060: NOTIFY sip:10277x...@192.168.233.237:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.233.235:5060;branch=z9hG4bK5cb290bd Max-Forwards: 70 From: Robyns sip:1...@192.168.233.235:5060;tag=as405c743f To: Ira sip:10277x...@192.168.233.235:5060;tag=90f0be2565982a7 Contact: sip:1...@192.168.233.235 Call-ID: 5381b179eb2eec40ffa7dc855e671...@192.168.233.237 CSeq: 106 NOTIFY User-Agent: Asterisk PBX 1.6.2.0-beta3 Event: dialog
[asterisk-users] Truecall
This has to be an Asterisk based appliance no? http://www.truecall.co.uk/acatalog/trueCall_Features.html Looks pretty easy to setup using AstLinux or similar. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
