[asterisk-users] How does leastrecent work?
Hi, Could someone kindly explain how does least recent strategy work? According to the config: leastrecent: rings the interface that least recently received a call That does not explain much in detail. What happen if agent been idle (pause member) or in wrap for length of time.. how does leastrecent take this into account? Thanks! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Auto Answer or AgentLogin stay on the line?
Hi, I am in process developing Multi-Tenant system for Call Centers. I am considering what are the best option for Agent to Login and and wait for the calls from the Queue. Option 1: AgentLogin (staying on the line with music on hold and bridging the call when a customer enters the queue) Option 2: After AgentLogin then enable Auto-Answer feature on the SIP Phones. I am considering going for Option 1 because it will work on any sip phones but it might be impossible to use AgentLogin if calls are distributed to many queues on multiple servers and leastrecent would not work? Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Distributed Device State options
Hi, What options do I have to setup Distributed Device State across to multiple Asterisk Servers? If an agent is on the phone on a queue on one of the Asterisk server, other servers will need to about it and therefore, will be able to operate adequately. For instance, an agent is a member of two queues (app_queue) and those queues on separate server. I have been reading about Distributed Device State using Corosync or XMPP. Is there any other options but reliable? How about share device state through Redis by looking at the device state on Agent/Queue. This could be done in Lua in the dialplan. Thanks Shahid -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Live Recording on the NAS?
Hi, I am planning to move Asterisk from physical server to a VM on a ESXi host. VMware datastore / VM's will be stored on the shared storage on the NAS (NSF). I might get Synology NAS. Do you store call live recording on the NAS? There would be around 60 concurrent calls recording at the same time and it may cause network bottleneck. There will be other VM's stored on the NAS like Windows Servers, Linux Servers, Database, etc. Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AgentLogin() on the multiple servers?
You say AgentLogin() must apply to all servers? but agent will be already on the line from a single server. When agent stays on the line after logging in, that channel will be only on one Asterisk server? If there is an incoming call getting routed on another Asterisk server, that agent and the incoming call will need to be bridged together? On Tue, Sep 15, 2015 at 11:24 AM, Faheem Muhammad <faheem2...@gmail.com> wrote: > You can achieve this by choosing one of asterisk server for pins > collection on extension 1234. When any member/extension dial that extension > you need to call a script that will make AMI connection on all servers and > do AgentLogin/QueueAdd Request. > You need to do ami login and call the AMI request QueueAdd on all server > where you have define different queues. It will make the agent login on all > Queue servers. > Below is snippet for making QueueAdd request from AMI. > > - > > Action: QueueAdd > Queue: supportqueue > Interface: sip/1122 > Penalty: 1 > > > Regards, > Muhammad Faheem > > > On Tue, Sep 15, 2015 at 3:46 AM, Shahid H <shah...@gmail.com> wrote: > >> Hello, >> >> Let say all the SIP devices will be registered on the proxy like kamailio. >> >> Agent is a member of Support and Billings Queues on the asterisk servers. >> Support queue on "Server A" and Billings Queue on "Server B" for example. >> This will be done via RealTime Queue. >> >> I want Agent to dial 1234 on a sip device and it will prompt to enter a >> pin number to Login via AgentLogin(). Agent will stay on the line after >> logged in and wait for the calls.. I understand how this work from single >> asterisk server. >> >> But how is it possible for Agent to stay on the line from multiple >> asterisk servers or how it should be done? If agent dial 1245 for logging >> in - does kamailio randomly need to pink any server and then prompt for Pin >> via AgentLogin()? >> >> Thanks >> >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >>http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >>http://lists.digium.com/mailman/listinfo/asterisk-users >> > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AgentLogin() on the multiple servers?
Hello, Let say all the SIP devices will be registered on the proxy like kamailio. Agent is a member of Support and Billings Queues on the asterisk servers. Support queue on "Server A" and Billings Queue on "Server B" for example. This will be done via RealTime Queue. I want Agent to dial 1234 on a sip device and it will prompt to enter a pin number to Login via AgentLogin(). Agent will stay on the line after logged in and wait for the calls.. I understand how this work from single asterisk server. But how is it possible for Agent to stay on the line from multiple asterisk servers or how it should be done? If agent dial 1245 for logging in - does kamailio randomly need to pink any server and then prompt for Pin via AgentLogin()? Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Looking for Asterisk Consultants & Experts
Hello, Can someone recommend me where is best place to find Asterisk Expert/Consultant for freelance work? If you are interested to work as a freelancer, you can email me directly. Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AgentRequest() and which agent id?
Thank you. I will test it today.
Is it possible to build a list of agent-id in MySQL Database rather than
agent.conf?
I am not sure what is the best approach, maybe you can give me some
direction. On the backend (web based) Admin can create a new Agent for
agent web panel login.
Admin will enter their new username and password for web login. Admin also
need to select a agent-id for a new agent.
I have two options.
- Build hundreds static agent-id in agents.conf
- Dynamic agent-id in mysql table (Not associated with agent.conf). Is this
possible?
Thanks
On Fri, Aug 7, 2015 at 4:50 PM, Richard Mudgett rmudg...@digium.com wrote:
On Fri, Aug 7, 2015 at 10:06 AM, Shahid H shah...@gmail.com wrote:
Hi,
If agents is already logged in via AgentLogin() and users dialled
extension 300 which will be placed in Queue(support-queue).
How to find out which agent is available I can put their Agent id
in AgentRequest() ?
If this is not a good approach then how it should be done?
Agent should automatically get next call when he/she is available.
extensions.conf
[LocalSets]
exten = 300,1,Answer()
same = n,Queue(support-queue)
same = n,hangup
[agents] ;callback
exten = 800,1,AgentRequest() ; How to get Agent id that is available
to take call?
Replace above line with the following line if your agent-ids are 4 digit
numeric:
exten = _,1,AgentRequest(${EXTEN})
same = n,NoOp(AGENT_STATUS is ${AGENT_STATUS})
same = n,Playback(silence/1)
same = n,Hangup()
Example in the queue.conf
member = Local/800@agents,0,Name 1,Agent:1001
Replace the above line for agent 1001 with the following:
member = Local/1001@agents,0,Name,1,Agent:1001
For other agents follow the similar pattern:
member = Local/agent-id@agents,0,Name,1,Agent:agent-id
Richard
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[asterisk-users] AgentRequest() and which agent id?
Hi,
If agents is already logged in via AgentLogin() and users dialled extension
300 which will be placed in Queue(support-queue).
How to find out which agent is available I can put their Agent id
in AgentRequest() ?
If this is not a good approach then how it should be done?
Agent should automatically get next call when he/she is available.
extensions.conf
[LocalSets]
exten = 300,1,Answer()
same = n,Queue(support-queue)
same = n,hangup
[agents] ;callback
exten = 800,1,AgentRequest() ; How to get Agent id that is available
to take call?
same = n,NoOp(AGENT_STATUS is ${AGENT_STATUS})
same = n,Playback(silence/1)
same = n,Hangup()
Example in the queue.conf
member = Local/800@agents,0,Name 1,Agent:1001
and more..
Thanks
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[asterisk-users] Buzzing / Humming Noise
Hi Lately I've been having problem with the handsets on the polycom phones. The agents are complaining that the humming/buzzing can be heard on random days. When the agents touch their computer the buzzing may increase louder or reduced. Also when they grab the handset cable, the buzzing noise is less loud and is loud again when I let it go. I have tried changing the handsets, problem not fixed. What is causing this? Is this networking issue? I have tried restarting the PBX Servers, switches, etc. Still same problem. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Live Recording on the Storage Server?
Hello, I am wondering has anyone used Live Recording (monitor or mixmonitor) on to Storage Server via network 1 Gigabit connection? Does it perform well, let say about 50 live recordings at the same time. I am planning to make some system changes at work. I would like to put Asterisk VM on a ESXi host and the datastore will be hosted on Storage Server. On a ESXi host, there will be a few VM's: Asterisk VM Windows Server VM Linux Web Server VM Windows 7 VM What I am concern that users on the workstations will browse their files (home shares) and it may interrupt asterisk live recording because it is shared on the same Storage Server? Cheers. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk AMI - PHP or Node.js?
I would like to develop a Call Center Dialer (outbound and inbound calls) and it would use AMI method to communicate with Asterisk Server. A daemon would need to run in the background, would you recommend coding in PHP or Node.js? which would be much faster and stable. Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk AMI - PHP or Node.js?
Thanks Daniel, that was useful, I will check those links :) I am pretty good with PHP and jQuery. So I guess learning Node.js shouldn't be too difficult. If I decided to use Node.js - what is the best way to communicate with a browser to AMI process? Send a XML or HTTP command from a browser to AMI process .. or whatever I execute on the browser - it save the commands to the database.. a process will listen the commands from a database. Other options are http requests and do the ipc... or maybe WebSocket? On Sat, Dec 28, 2013 at 11:41 AM, Daniel Jenkins dan.jenkin...@gmail.comwrote: On Sat, Dec 28, 2013 at 11:32 AM, Shahid H shah...@gmail.com wrote: Hi, I would like to develop a Call Center Dialer (outbound and inbound calls) and it would use AMI method to communicate with Asterisk Server. A daemon would need to run in the background, would you recommend coding in PHP or Node.js? which would be much faster and stable. I've done something very similar, and I've done it with Node.js - it's a great fit for it; whereas you're kinda forcing PHP to do something it wasn't 100% designed for; in my experience Node is a much better fit for it. I've spoken at Astricon about doing it and it's fairly easy with modules that already exist in the Node.js ecosystem. The talk I did in 2012 is probably the most relevant: http://www.youtube.com/watch?v=FCwzbgR_adQ The slides can be found here: http://www.slideshare.net/danjenkins/asterisk-html5-and-nodejs-a-world-of-endless-possibilities-14881614 And my module for talking to asterisk via the AMI is available on npm https://npmjs.org/package/asterisk-ami On the question of stability/speed, I don't know to be honest but I would guess that Node would be faster due to it's async nature; but whether you would really see much of that speed gain due to async is entirely dependant on your application and how it runs, what it talks to, are any of these things potentially blocking. Node.js is nearing V1 but to be honest, it's the most stable it's ever been, with HUGE companies using it in production environments. The system that I wrote is still running on Node v 0.6 and has been up and running with near 0 downtime for over a year, and the reasons for downtime are external, such as server moves, DNS changes etc. Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Let me know if you want to know any more. Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk AMI - Create a daemon (background process)
I wanted to create a daemon (background process) in PHP. A daemon will use socket to connect with Asterisk AMI to send events and listen the actions. A daemon will also listen the commands from agents via HTTP, for example: A agent pressed a hang up button on a browser - it will send http command to a daemon. A daemon received a command and will then send Hang Up Action to AMI. How should a daemon process be designed to listen multiple actions and events? For example: 50 agents currently on the calls and how should a daemon to monitor the Actions/Events from 50 agents? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk AMI - daemon process
I wanted to create a daemon (background process) in PHP. A daemon will use socket to connect with Asterisk AMI to send events and listen the actions. A daemon will also listen the commands from agents via HTTP, for example: A agent pressed a hang up button on a browser - it will send http command to a daemon. A daemon received a command and will then send Hang Up Action to AMI. How should a daemon process be designed to listen multiple actions and events? For example: 50 agents currently on the calls and how should a daemon to monitor the Actions/Events from 50 agents? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Suggestion of Server Specifications for Asterisk
I have bought a new server today: i7-2600 CPU, 8GB and 2 x 256GB SSDs. 100Mbit Connection. I hope CPU is powerful enough for 200 concurrent calls. On Sun, Aug 5, 2012 at 1:57 AM, Michelle Dupuis mdup...@ocg.ca wrote: That's how we do it - write to a memory based (ramdisk) disk then write to HDD upon call completion. We haven't tried a SSD but that may be necessary depending on your call volumes. -- *From:* asterisk-users-boun...@lists.digium.com [ asterisk-users-boun...@lists.digium.com] On Behalf Of Ryan Wagoner [ rswago...@gmail.com] *Sent:* Saturday, August 04, 2012 7:34 PM *To:* Asterisk Users List *Subject:* Re: [asterisk-users] Suggestion of Server Specifications for Asterisk On Sat, Aug 4, 2012 at 1:22 PM, Shahid H shah...@gmail.com wrote: Instead of buying expensive disk.. I might setup a ramdisk (about 2GB) to do 200 calls recordings. Once the call hangup/completed it will then move recording file to SATA HDD. What do you think of this? You want some form of raid for redundancy. I usually go with two 15K SAS drives in raid 1 or four 7.2k SATA drives in raid 10. Performance between the two should be similar. With drives being as cheap as they are skip raid 5. Ryan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Suggestion of Server Specifications for Asterisk
Would a SSD drive be enough or do I need like Raid 10 (4 hard drives)? On Sat, Aug 4, 2012 at 10:17 AM, Leandro Dardini ldard...@gmail.com wrote: The busiest server I am managing reaches 120 concurrent channels (with mixed recording). It is a dual processor, dual core Intel 5150 with 16 GB of ram and raid sas controller. The load reaches rarely 3.0. Having to double the number of channels and due to the 100% call recordings, I'll go with a 16 cores. Memory will not a big issue and so the disk. 64kbit/s x 200 (even adding the overhead of the SIP and IP) will be under 20 Mbit/s, so a 100 Mbit/s will be fine. About UK provider, I can't be of any help... I know very good providers in Germany and Canada, where I am laying my servers, but none in UK. Leandro 2012/8/4 Shahid H shah...@gmail.com What the minimum Server Specifications do I need to run 200 concurrent channels at the time with .WAV recording (MixMonitor)? It will be connected via VOIP sip account. Codec will be ulaw. Which UK dedicated server provider do you recommend and how much bandwidth do I need? Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Suggestion of Server Specifications for Asterisk
Ahh I see. So I might as well get a normal sata disk? I thought I/O will be Bottleneck as well because 200 channels WAV recordings to disk at the same time. Which intel model 16 cores do you recommend? how about 12 cores? Thanks! On Sat, Aug 4, 2012 at 1:19 PM, Leandro Dardini ldard...@gmail.com wrote: It is not necessary to use an high performance drive. The bottleneck will be the processor, not the disk. A single disk can handle ten times the load of 200 ulaw channels. Leandro Il giorno 04/ago/2012 12:39, Shahid H shah...@gmail.com ha scritto: Would a SSD drive be enough or do I need like Raid 10 (4 hard drives)? On Sat, Aug 4, 2012 at 10:17 AM, Leandro Dardini ldard...@gmail.comwrote: The busiest server I am managing reaches 120 concurrent channels (with mixed recording). It is a dual processor, dual core Intel 5150 with 16 GB of ram and raid sas controller. The load reaches rarely 3.0. Having to double the number of channels and due to the 100% call recordings, I'll go with a 16 cores. Memory will not a big issue and so the disk. 64kbit/s x 200 (even adding the overhead of the SIP and IP) will be under 20 Mbit/s, so a 100 Mbit/s will be fine. About UK provider, I can't be of any help... I know very good providers in Germany and Canada, where I am laying my servers, but none in UK. Leandro 2012/8/4 Shahid H shah...@gmail.com What the minimum Server Specifications do I need to run 200 concurrent channels at the time with .WAV recording (MixMonitor)? It will be connected via VOIP sip account. Codec will be ulaw. Which UK dedicated server provider do you recommend and how much bandwidth do I need? Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Suggestion of Server Specifications for Asterisk
Instead of buying expensive disk.. I might setup a ramdisk (about 2GB) to do 200 calls recordings. Once the call hangup/completed it will then move recording file to SATA HDD. What do you think of this? On Sat, Aug 4, 2012 at 5:51 PM, Benny Amorsen benny+use...@amorsen.dkwrote: Leandro Dardini ldard...@gmail.com writes: A single sata disk will be an unacceptable single point of failure. Get three disks and get in raid5 configuration. You'll gain in safety and speed. RAID-5 is slower than single disks when it comes to write IOPS (a commit is not done until the slowest disk has answered). Avoid it for write heavy workloads at all costs unless you are writing sequentially in one file with write caching enabled. /Benny -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Suggestion of Server Specifications for Asterisk
What the minimum Server Specifications do I need to run 200 concurrent channels at the time with .WAV recording (MixMonitor)? It will be connected via VOIP sip account. Codec will be ulaw. Which UK dedicated server provider do you recommend and how much bandwidth do I need? Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 50% of time SendDTMF failed
I am having a problem with SendDTMF() - 50% of time it did not succeed. I suspect it is not sending clear DTMF tones to the IVR. For example: SendDTMF(w3w2ww1w4) Sometime digit 3 and 2 work, and failed to do digit 1. Sometime digit 3 work and failed to do number 2. Sometime all went through fine. dtmfmode=rfc2833 are set in the sip.conf file How do I debug to see what went wrong and how to fix? Asterisk 1.8.12.0 Installed on VPS (XEN, CentOS 5.x, 768 MB Ram, 1000 GB B/W - Located in UK) VOIP Provider in UK. Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] GSM gateway or PCI Card recommendation?
I am looking for a GSM Gateway or GSM PCI Card with minimum of 6 Sim Cards slots. Which one do you recommend and easier to setup? As long it work on UK mobile network and make 6 calls simultaneously. Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] GSM gateway or PCI Card recommendation?
I am looking for a GSM Gateway or GSM PCI Card with minimum of 6 Sim Cards slots. Which one do you recommend and easier to setup? As long it work on UK mobile network and make 6 calls simultaneously. Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Event response (AMI)
When I execute the ACTION commands set then the EVENT would response back. How would I know which ACTION are they belong/reference to? For example: ACTION: Originate Channel: SIP/test Exten: 215 Timeout: 3 Context: test Priority: 1 ActionID: 1333 Response: Success ActionID: 1333 Message: Originate successfully queued EVENT response when I hang up the call: Event: Hangup Privilege: call,all Channel: SIP/test-007f Uniqueid: 1336690030.189 CallerIDNum: unknown CallerIDName: unknown Cause: 16 Cause-txt: Normal Clearing As you can see, how would I know which which ACTION was that belong to? If I were coding in PHP (AMI) to Originate the calls then I want to detect which call hanged up. Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Event response (AMI)
Thanks for the detailed information.
I have came up with this solution:
I've generate random CALLID and assign to UserEvent()
exten = xxx,1,UserEvent(CallStatus,UniqueID: ${UNIQUEID}, CallID:
${CALLID})
I then find matching CALLID from the EVENT response which will also tell me
what the UniqueID.
What do you think of this solution?
Thanks
On Fri, May 11, 2012 at 2:31 PM, Matthew Jordan mjor...@digium.com wrote:
- Original Message -
From: Shahid H shah...@gmail.com
To: asterisk-users@lists.digium.com
Sent: Friday, May 11, 2012 6:12:25 AM
Subject: [asterisk-users] Event response (AMI)
When I execute the ACTION commands set then the EVENT would response
back. How would I know which ACTION are they belong/reference to?
For example:
ACTION: Originate
Channel: SIP/test
Exten: 215
Timeout: 3
Context: test
Priority: 1
ActionID: 1333
Response: Success
ActionID: 1333
Message: Originate successfully queued
EVENT response when I hang up the call:
Event: Hangup
Privilege: call,all
Channel: SIP/test-007f
Uniqueid: 1336690030.189
CallerIDNum: unknown
CallerIDName: unknown
Cause: 16
Cause-txt: Normal Clearing
As you can see, how would I know which which ACTION was that belong
to?
If I were coding in PHP (AMI) to Originate the calls then I want to
detect which call hanged up.
Thanks
In general, there is no explicit tying of events with a particular action
that caused those events, as AMI events are completely independent of AMI
actions. The exception to this, of course, are AMI events that are
reporting
success/failure of an AMI action, in which case they will share the same
ActionId header (if the action specified an ActionId as well).
In your particular case, if I were writing a system that wanted to
associate
a created channel with an Originate Action, after I issue the Originate,
I'd listen for a NewChannel event. If that NewChannel event specified a
channel that was created in the context I specified and with a
technology/extension that I specified, I'd set that as the channel I just
asked to be created. From there on, subsequent events (VarSet, NewExten,
Hangup, etc.) that are associated with that channel will contain a Channel:
header with that value.
--
Matthew Jordan
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com http://asterisk.org
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[asterisk-users] Event response (AMI)
When I execute the Action commands set then the Event would response back. How would I know which Action are they belong/reference to? For example: ACTION: Originate Channel: SIP/test Exten: 215 Timeout: 3 Context: test Priority: 1 ActionID: 1333 Response: Success ActionID: 1333 Message: Originate successfully queued Event response when I hang up the call: Event: Hangup Privilege: call,all Channel: SIP/test-007f Uniqueid: 1336690030.189 CallerIDNum: unknown CallerIDName: unknown Cause: 16 Cause-txt: Normal Clearing As you can see, how would I know which which ACTION was that belong to? If I were coding in PHP (AMI) to Originate the calls then I want to detect which call hanged up. Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Why do I get call twice in one go?
I understand why do I get call twice to my mobile when I execute the following AMI command sets: ACTION: Originate Channel: Local/800@test Timeout: 6 Priority: 1 and my dialplan look like this: [test] exten = 800,1,DIAL(SIP/447xx@voip); exten = 800,n,Hangup() How to prevent getting called twice in one go when I execute this AMI command? Thanks... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Why did it Hangup?
I am learning how to use AMI and I am having 1 problem.. When I make a call
to my mobile phone and when I answer it - it get disconnected/hangup right
away.
Why is that? What is the solution to stop that?
For example:
ACTION: Originate
Channel: SIP/447XXX@vpsprovider
Exten: 210
Priority: 1
CallerID: 0044123456789
Timeout: 6
Context: test
exten = 210,1,Answer
exten =
210,n,Set(MONITOR_FILENAME=Record-${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)})
exten = 210,n,SendDTMF(2w3w)
exten = 210,n,Monitor(wav,${MONITOR_FILENAME},ib)
exten = 210,n,Hangup()
Before I had Dial() in the dialplan and it work great and no hangup. Now I
am using AMI method.
Thanks
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Re: [asterisk-users] Why did it Hangup?
No, that 'timeout' option is when I don't answer the call.
My problem is when I DO answer the call, it get disconnected right away.
Yes hangup() get executed right away when I answer the call.
On Tue, May 8, 2012 at 9:17 PM, Danny Nicholas da...@debsinc.com wrote:
It is likely the 60 second timeout you are providing. Or it could be the
hangup() command in the 210 context.
** **
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Shahid H
*Sent:* Tuesday, May 08, 2012 3:11 PM
*To:* asterisk-users@lists.digium.com
*Subject:* [asterisk-users] Why did it Hangup?
** **
I am learning how to use AMI and I am having 1 problem.. When I make a
call to my mobile phone and when I answer it - it get disconnected/hangup
right away.
** **
Why is that? What is the solution to stop that?
** **
For example:
** **
ACTION: Originate
Channel: SIP/447XXX@vpsprovider
Exten: 210
Priority: 1
CallerID: 0044123456789
Timeout: 6
Context: test
** **
** **
exten = 210,1,Answer
exten =
210,n,Set(MONITOR_FILENAME=Record-${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)})***
*
exten = 210,n,SendDTMF(2w3w)
exten = 210,n,Monitor(wav,${MONITOR_FILENAME},ib)
exten = 210,n,Hangup()
** **
** **
Before I had Dial() in the dialplan and it work great and no hangup. Now
I am using AMI method.
** **
** **
Thanks
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Re: [asterisk-users] Why did it Hangup?
I have tried that and that did not fixed the problem,
However, I have added this in the dialplan:
exten = 210,n,Wait(60)
That will hangup the call after 60 seconds...
That is fine by me but now Monitor() dont even work now... it does not
record a call...?
On Tue, May 8, 2012 at 9:22 PM, Danny Nicholas da...@debsinc.com wrote:
Since you are Originating the call, the hangup command isn’t needed.
Remove and reload.
** **
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Shahid H
*Sent:* Tuesday, May 08, 2012 3:20 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Why did it Hangup?
** **
No, that 'timeout' option is when I don't answer the call.
** **
My problem is when I DO answer the call, it get disconnected right away.**
**
** **
Yes hangup() get executed right away when I answer the call.
** **
** **
On Tue, May 8, 2012 at 9:17 PM, Danny Nicholas da...@debsinc.com wrote:*
***
It is likely the 60 second timeout you are providing. Or it could be the
hangup() command in the 210 context.
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Shahid H
*Sent:* Tuesday, May 08, 2012 3:11 PM
*To:* asterisk-users@lists.digium.com
*Subject:* [asterisk-users] Why did it Hangup?
I am learning how to use AMI and I am having 1 problem.. When I make a
call to my mobile phone and when I answer it - it get disconnected/hangup
right away.
Why is that? What is the solution to stop that?
For example:
ACTION: Originate
Channel: SIP/447XXX@vpsprovider
Exten: 210
Priority: 1
CallerID: 0044123456789
Timeout: 6
Context: test
exten = 210,1,Answer
exten =
210,n,Set(MONITOR_FILENAME=Record-${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)})***
*
exten = 210,n,SendDTMF(2w3w)
exten = 210,n,Monitor(wav,${MONITOR_FILENAME},ib)
exten = 210,n,Hangup()
Before I had Dial() in the dialplan and it work great and no hangup. Now
I am using AMI method.
Thanks
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** **
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[asterisk-users] How to hang up a call after sending SendDTMF() ?
When SendDTMF() finish the process then I want to hang up the call after 20 seconds.. What is the solution to do this? I know there is S(x) option for Dial() application but it still count during SendDTMF() process. Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Why SendDTMF is not working?
Hello, I am having a problem with SendDTMF - it is not sending the numbers properly during the phone call.. I want the numbers key to to be pressed/sent automatically after 3 seconds during a phone call. I use software phone to test it... when I dialed 501, I cant hear anything for about 10 seconds (this is because of SendDTMF) and then I can hear the operator saying to enter the numbers but SendDTMF already did it?! Asterisk server are connected to voip.ms provider. I have spent many hours trying to get to work, how to fix this issue? See the configuration and debug log below: extensions.conf [test] exten = 501,1,Set(CALLERID(num)=004471XXX) exten = 501,n,Dial(SIP/+44797XX@voipms,30,M(sendnumber)t) exten = 501,n,Hangup() [macro-sendnumber] exten = s,1,Wait(3) exten = s,n,SendDTMF(www0w7w8w8wXwXwXwXwXwX) sip.conf == [general] context=default tcpbindaddr=0.0.0.0 dtmfmode = rfc2833 register = x:vxx...@london.voip.ms:5060 [test] type=peer secret=2xxx host=dynamic context=test [voipms] canreinvite=no host=london.voip.ms secret=xx type=peer username=135xxx ;your account disallow=all allow=gsm ; allow=g729 ; Uncomment if you support G729 fromuser=135xxx insecure=invite trustrpid=yes sendrpid=yes nat=yes dtmfmode=rfc2833 debug: = == Using SIP RTP CoS mark 5 -- Executing [501@test:1] Set(SIP/test-0026, CALLERID(num)=004471XX) in new stack -- Executing [501@test:2] Dial(SIP/test-0026, SIP/+4479XX@voipms,30,M(sendnumber)t) in new stack == Using SIP RTP CoS mark 5 -- Called +44797XX@voipms -- SIP/voipms-0027 is making progress passing it to SIP/test-0026 -- SIP/voipms-0027 answered SIP/test-0026 -- Executing [s@macro-sendnumber:1] Wait(SIP/voipms-0027, 3) in new stack -- Executing [s@macro-sendnumber:2] SendDTMF(SIP/voipms-0027, www0w7w8wXwXwXwXw4wXwXwX) in new stack If you need any more information or debug, let me know! Thanks! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Why SendDTMF is not working?
Thanks for the suggestion Markus. Here what I did: In the logger.config I have added 'dtmf': console = notice,warning,error,dtmf and then in sip.conf: allow=ulaw allow=alaw ; allow=gsm dtmfmode=inband I've added a test to call my mobile: exten = 123,1,Dial(SIP/+4477XXX@voipms,,D(1ww2ww3ww4)) exten = 123,n,Hangup() then restarted asterisk and logged into console (asterisk -r) I've call my mobile using softphone, I did not see 1,2,3,4 digits being sent on the console but I can hear broken/unclear DTMF on the mobile... however when I press digits on the softphone I can hear DTMF clear how it should be on my mobile and on the console it is showing DTMF: astrisk*CLI [May 6 14:13:06] DTMF[28559]: channel.c:3082 __ast_read: DTMF begin '4' received on SIP/test-001c [May 6 14:13:06] DTMF[28559]: channel.c:3092 __ast_read: DTMF begin passthrough '4' on SIP/test-001c [May 6 14:13:06] DTMF[28559]: channel.c:2997 __ast_read: DTMF end '4' received on SIP/test-001c, duration 120 ms [May 6 14:13:06] DTMF[28559]: channel.c:3037 __ast_read: DTMF end accepted with begin '4' on SIP/test-001c [May 6 14:13:06] DTMF[28559]: channel.c:3066 __ast_read: DTMF end passthrough '4' on SIP/test-001c [May 6 14:13:07] DTMF[28559]: channel.c:3082 __ast_read: DTMF begin '5' received on SIP/test-001c [May 6 14:13:07] DTMF[28559]: channel.c:3092 __ast_read: DTMF begin passthrough '5' on SIP/test-001c [May 6 14:13:07] DTMF[28559]: channel.c:2997 __ast_read: DTMF end '5' received on SIP/test-001c, duration 120 ms [May 6 14:13:07] DTMF[28559]: channel.c:3037 __ast_read: DTMF end accepted with begin '5' on SIP/test-001c [May 6 14:13:07] DTMF[28559]: channel.c:3066 __ast_read: DTMF end passthrough '5' on SIP/test-001c [May 6 14:13:08] DTMF[28559]: channel.c:3082 __ast_read: DTMF begin '6' received on SIP/test-001c [May 6 14:13:08] DTMF[28559]: channel.c:3092 __ast_read: DTMF begin passthrough '6' on SIP/test-001c [May 6 14:13:08] DTMF[28559]: channel.c:2997 __ast_read: DTMF end '6' received on SIP/test-001c, duration 120 ms [May 6 14:13:08] DTMF[28559]: channel.c:3037 __ast_read: DTMF end accepted with begin '6' on SIP/test-001c [May 6 14:13:08] DTMF[28559]: channel.c:3066 __ast_read: DTMF end passthrough '6' on SIP/test-001c Thanks! On Sun, May 6, 2012 at 1:03 PM, Markus unive...@truemetal.org wrote: Am 06.05.2012 13:46, schrieb Shahid H: Hello, I am having a problem with SendDTMF - it is not sending the numbers properly during the phone call.. I want the numbers key to to be pressed/sent automatically after 3 seconds during a phone call. Log the actual DTMF to your console, set in logger.conf: console = something,something,dtmf Then try again and check if you see the actual DTMF. If you do and it still doesn't work, try dtmfmode=inband for your voipms peer. rfc2833 has been working always unreliable for me. Also, I'm doing DTMF like this: exten = 5000,n,Dial(SIP/123456@**provider,,D(ww1ww2ww3ww4)) Just use more w's to generate your 3 seconds pause. No need for SendDTMF. For more debugging just call yourself on your UK mobile from a softphone and press digits and watch the console and listen on your mobile if you hear the DTMF. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Why SendDTMF is not working?
When I changed back to dtmfmode=rfc2833 and I cant hear the DTMF sound.. completely silent. Indeed I have put disallow=all before the allow=ulaw allow=alaw sip show channels in the CLI show during a call: 78.129.xxx.xx +447715d909406db14d2 0x4 (ulaw) No Tx: ACK 94.192.xxx.xx test MTNlNGNkYjlhODA 0x4 (ulaw) No Rx: ACK Still no luck to get DTMF to work :( Thanks Shahid On Sun, May 6, 2012 at 2:54 PM, Eric Wieling ewiel...@nyigc.com wrote: Now you have a totally different issue. 8-) While the call is up do a sip show channels in the CLI. This will show you the ACTUAL codec for the call. Likely the call was still using GSM. Did you remember to put a disallow=all before the allow= lines? I recommend dtmfmode=rfc2833 with whatever codec you want to use. Inband DTMF will sound broken and distorted if it is sent over most codecs. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of Shahid H Sent: Sunday, May 06, 2012 9:16 AM To: Markus Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Why SendDTMF is not working? Thanks for the suggestion Markus. Here what I did: In the logger.config I have added 'dtmf': console = notice,warning,error,dtmf and then in sip.conf: allow=ulaw allow=alaw ; allow=gsm dtmfmode=inband I've added a test to call my mobile: exten = 123,1,Dial(SIP/+4477XXX@voipms,,D(1ww2ww3ww4)) exten = 123,n,Hangup() then restarted asterisk and logged into console (asterisk -r) I've call my mobile using softphone, I did not see 1,2,3,4 digits being sent on the console but I can hear broken/unclear DTMF on the mobile... however when I press digits on the softphone I can hear DTMF clear how it should be on my mobile and on the console it is showing DTMF: astrisk*CLI [May 6 14:13:06] DTMF[28559]: channel.c:3082 __ast_read: DTMF begin '4' received on SIP/test-001c [May 6 14:13:06] DTMF[28559]: channel.c:3092 __ast_read: DTMF begin passthrough '4' on SIP/test-001c [May 6 14:13:06] DTMF[28559]: channel.c:2997 __ast_read: DTMF end '4' received on SIP/test-001c, duration 120 ms [May 6 14:13:06] DTMF[28559]: channel.c:3037 __ast_read: DTMF end accepted with begin '4' on SIP/test-001c [May 6 14:13:06] DTMF[28559]: channel.c:3066 __ast_read: DTMF end passthrough '4' on SIP/test-001c [May 6 14:13:07] DTMF[28559]: channel.c:3082 __ast_read: DTMF begin '5' received on SIP/test-001c [May 6 14:13:07] DTMF[28559]: channel.c:3092 __ast_read: DTMF begin passthrough '5' on SIP/test-001c [May 6 14:13:07] DTMF[28559]: channel.c:2997 __ast_read: DTMF end '5' received on SIP/test-001c, duration 120 ms [May 6 14:13:07] DTMF[28559]: channel.c:3037 __ast_read: DTMF end accepted with begin '5' on SIP/test-001c [May 6 14:13:07] DTMF[28559]: channel.c:3066 __ast_read: DTMF end passthrough '5' on SIP/test-001c [May 6 14:13:08] DTMF[28559]: channel.c:3082 __ast_read: DTMF begin '6' received on SIP/test-001c [May 6 14:13:08] DTMF[28559]: channel.c:3092 __ast_read: DTMF begin passthrough '6' on SIP/test-001c [May 6 14:13:08] DTMF[28559]: channel.c:2997 __ast_read: DTMF end '6' received on SIP/test-001c, duration 120 ms [May 6 14:13:08] DTMF[28559]: channel.c:3037 __ast_read: DTMF end accepted with begin '6' on SIP/test-001c [May 6 14:13:08] DTMF[28559]: channel.c:3066 __ast_read: DTMF end passthrough '6' on SIP/test-001c Thanks! On Sun, May 6, 2012 at 1:03 PM, Markus unive...@truemetal.org wrote: Am 06.05.2012 13:46, schrieb Shahid H: Hello, I am having a problem with SendDTMF - it is not sending the numbers properly during the phone call.. I want the numbers key to to be pressed/sent automatically after 3 seconds during a phone call. Log the actual DTMF to your console, set in logger.conf: console = something,something,dtmf Then try again and check if you see the actual DTMF. If you do and it still doesn't work, try dtmfmode=inband for your voipms peer. rfc2833 has been working always unreliable for me. Also, I'm doing DTMF like this: exten = 5000,n,Dial(SIP/123456@provider,,D(ww1ww2ww3ww4)) Just use more w's to generate your 3 seconds pause. No need for SendDTMF. For more debugging just call yourself on your UK mobile from a softphone and press digits and watch the console and listen on your mobile if you hear the DTMF. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org
Re: [asterisk-users] Why SendDTMF is not working?
Here is another debug log: == Using SIP RTP CoS mark 5 -- Executing [123@test2:1] Dial(SIP/test2-0008, SIP/+44776@voipms,,D(1ww2ww3ww4)) in new stack == Using SIP RTP CoS mark 5 -- Called SIP/+44776XX@voipms -- SIP/voipms-0009 is making progress passing it to SIP/test2-0008 -- SIP/voipms-0009 answered SIP/test2-0008 -- Sending DTMF '1ww2ww3ww4' to the called party. -- Locally bridging SIP/test2-0008 and SIP/voipms-0009 When DTMF is finish then Locally bridging is executed... On the softphone it say State: Early Media while it sending DTMF even though I cant hear DTMF sound.. after 10 seconds State changed to Up (I can hear talking to myself). On Sun, May 6, 2012 at 4:18 PM, Shahid H shah...@gmail.com wrote: When I changed back to dtmfmode=rfc2833 and I cant hear the DTMF sound.. completely silent. Indeed I have put disallow=all before the allow=ulaw allow=alaw sip show channels in the CLI show during a call: 78.129.xxx.xx +447715d909406db14d2 0x4 (ulaw) No Tx: ACK 94.192.xxx.xx test MTNlNGNkYjlhODA 0x4 (ulaw) No Rx: ACK Still no luck to get DTMF to work :( Thanks Shahid On Sun, May 6, 2012 at 2:54 PM, Eric Wieling ewiel...@nyigc.com wrote: Now you have a totally different issue. 8-) While the call is up do a sip show channels in the CLI. This will show you the ACTUAL codec for the call. Likely the call was still using GSM. Did you remember to put a disallow=all before the allow= lines? I recommend dtmfmode=rfc2833 with whatever codec you want to use. Inband DTMF will sound broken and distorted if it is sent over most codecs. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of Shahid H Sent: Sunday, May 06, 2012 9:16 AM To: Markus Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Why SendDTMF is not working? Thanks for the suggestion Markus. Here what I did: In the logger.config I have added 'dtmf': console = notice,warning,error,dtmf and then in sip.conf: allow=ulaw allow=alaw ; allow=gsm dtmfmode=inband I've added a test to call my mobile: exten = 123,1,Dial(SIP/+4477XXX@voipms,,D(1ww2ww3ww4)) exten = 123,n,Hangup() then restarted asterisk and logged into console (asterisk -r) I've call my mobile using softphone, I did not see 1,2,3,4 digits being sent on the console but I can hear broken/unclear DTMF on the mobile... however when I press digits on the softphone I can hear DTMF clear how it should be on my mobile and on the console it is showing DTMF: astrisk*CLI [May 6 14:13:06] DTMF[28559]: channel.c:3082 __ast_read: DTMF begin '4' received on SIP/test-001c [May 6 14:13:06] DTMF[28559]: channel.c:3092 __ast_read: DTMF begin passthrough '4' on SIP/test-001c [May 6 14:13:06] DTMF[28559]: channel.c:2997 __ast_read: DTMF end '4' received on SIP/test-001c, duration 120 ms [May 6 14:13:06] DTMF[28559]: channel.c:3037 __ast_read: DTMF end accepted with begin '4' on SIP/test-001c [May 6 14:13:06] DTMF[28559]: channel.c:3066 __ast_read: DTMF end passthrough '4' on SIP/test-001c [May 6 14:13:07] DTMF[28559]: channel.c:3082 __ast_read: DTMF begin '5' received on SIP/test-001c [May 6 14:13:07] DTMF[28559]: channel.c:3092 __ast_read: DTMF begin passthrough '5' on SIP/test-001c [May 6 14:13:07] DTMF[28559]: channel.c:2997 __ast_read: DTMF end '5' received on SIP/test-001c, duration 120 ms [May 6 14:13:07] DTMF[28559]: channel.c:3037 __ast_read: DTMF end accepted with begin '5' on SIP/test-001c [May 6 14:13:07] DTMF[28559]: channel.c:3066 __ast_read: DTMF end passthrough '5' on SIP/test-001c [May 6 14:13:08] DTMF[28559]: channel.c:3082 __ast_read: DTMF begin '6' received on SIP/test-001c [May 6 14:13:08] DTMF[28559]: channel.c:3092 __ast_read: DTMF begin passthrough '6' on SIP/test-001c [May 6 14:13:08] DTMF[28559]: channel.c:2997 __ast_read: DTMF end '6' received on SIP/test-001c, duration 120 ms [May 6 14:13:08] DTMF[28559]: channel.c:3037 __ast_read: DTMF end accepted with begin '6' on SIP/test-001c [May 6 14:13:08] DTMF[28559]: channel.c:3066 __ast_read: DTMF end passthrough '6' on SIP/test-001c Thanks! On Sun, May 6, 2012 at 1:03 PM, Markus unive...@truemetal.org wrote: Am 06.05.2012 13:46, schrieb Shahid H: Hello, I am having a problem with SendDTMF - it is not sending the numbers properly during the phone call.. I want the numbers key to to be pressed/sent automatically after 3 seconds during a phone call. Log the actual DTMF to your console, set in logger.conf: console = something,something,dtmf Then try again
Re: [asterisk-users] Why SendDTMF is not working?
Hey guys, I have managed to get to work Thanks for the help.. I just registered a new account at sipgate.co.uk and test it on asterisk... and DTMF worked well :) It seem voip.ms dont work well when sending DTMF to UK. Do anyone know UK/Europe voip provider to allow you change any callerID as you like without validation? I know voip.ms does it and sipgate don't allow it. Thanks! On Sun, May 6, 2012 at 5:08 PM, Shahid H shah...@gmail.com wrote: Here is another debug log: == Using SIP RTP CoS mark 5 -- Executing [123@test2:1] Dial(SIP/test2-0008, SIP/+44776@voipms,,D(1ww2ww3ww4)) in new stack == Using SIP RTP CoS mark 5 -- Called SIP/+44776XX@voipms -- SIP/voipms-0009 is making progress passing it to SIP/test2-0008 -- SIP/voipms-0009 answered SIP/test2-0008 -- Sending DTMF '1ww2ww3ww4' to the called party. -- Locally bridging SIP/test2-0008 and SIP/voipms-0009 When DTMF is finish then Locally bridging is executed... On the softphone it say State: Early Media while it sending DTMF even though I cant hear DTMF sound.. after 10 seconds State changed to Up (I can hear talking to myself). On Sun, May 6, 2012 at 4:18 PM, Shahid H shah...@gmail.com wrote: When I changed back to dtmfmode=rfc2833 and I cant hear the DTMF sound.. completely silent. Indeed I have put disallow=all before the allow=ulaw allow=alaw sip show channels in the CLI show during a call: 78.129.xxx.xx +447715d909406db14d2 0x4 (ulaw) No Tx: ACK 94.192.xxx.xx test MTNlNGNkYjlhODA 0x4 (ulaw) No Rx: ACK Still no luck to get DTMF to work :( Thanks Shahid On Sun, May 6, 2012 at 2:54 PM, Eric Wieling ewiel...@nyigc.com wrote: Now you have a totally different issue. 8-) While the call is up do a sip show channels in the CLI. This will show you the ACTUAL codec for the call. Likely the call was still using GSM. Did you remember to put a disallow=all before the allow= lines? I recommend dtmfmode=rfc2833 with whatever codec you want to use. Inband DTMF will sound broken and distorted if it is sent over most codecs. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of Shahid H Sent: Sunday, May 06, 2012 9:16 AM To: Markus Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Why SendDTMF is not working? Thanks for the suggestion Markus. Here what I did: In the logger.config I have added 'dtmf': console = notice,warning,error,dtmf and then in sip.conf: allow=ulaw allow=alaw ; allow=gsm dtmfmode=inband I've added a test to call my mobile: exten = 123,1,Dial(SIP/+4477XXX@voipms,,D(1ww2ww3ww4)) exten = 123,n,Hangup() then restarted asterisk and logged into console (asterisk -r) I've call my mobile using softphone, I did not see 1,2,3,4 digits being sent on the console but I can hear broken/unclear DTMF on the mobile... however when I press digits on the softphone I can hear DTMF clear how it should be on my mobile and on the console it is showing DTMF: astrisk*CLI [May 6 14:13:06] DTMF[28559]: channel.c:3082 __ast_read: DTMF begin '4' received on SIP/test-001c [May 6 14:13:06] DTMF[28559]: channel.c:3092 __ast_read: DTMF begin passthrough '4' on SIP/test-001c [May 6 14:13:06] DTMF[28559]: channel.c:2997 __ast_read: DTMF end '4' received on SIP/test-001c, duration 120 ms [May 6 14:13:06] DTMF[28559]: channel.c:3037 __ast_read: DTMF end accepted with begin '4' on SIP/test-001c [May 6 14:13:06] DTMF[28559]: channel.c:3066 __ast_read: DTMF end passthrough '4' on SIP/test-001c [May 6 14:13:07] DTMF[28559]: channel.c:3082 __ast_read: DTMF begin '5' received on SIP/test-001c [May 6 14:13:07] DTMF[28559]: channel.c:3092 __ast_read: DTMF begin passthrough '5' on SIP/test-001c [May 6 14:13:07] DTMF[28559]: channel.c:2997 __ast_read: DTMF end '5' received on SIP/test-001c, duration 120 ms [May 6 14:13:07] DTMF[28559]: channel.c:3037 __ast_read: DTMF end accepted with begin '5' on SIP/test-001c [May 6 14:13:07] DTMF[28559]: channel.c:3066 __ast_read: DTMF end passthrough '5' on SIP/test-001c [May 6 14:13:08] DTMF[28559]: channel.c:3082 __ast_read: DTMF begin '6' received on SIP/test-001c [May 6 14:13:08] DTMF[28559]: channel.c:3092 __ast_read: DTMF begin passthrough '6' on SIP/test-001c [May 6 14:13:08] DTMF[28559]: channel.c:2997 __ast_read: DTMF end '6' received on SIP/test-001c, duration 120 ms [May 6 14:13:08] DTMF[28559]: channel.c:3037 __ast_read: DTMF end accepted with begin '6' on SIP/test-001c [May 6 14:13:08] DTMF[28559]: channel.c:3066 __ast_read: DTMF end passthrough '6' on SIP/test-001c Thanks! On Sun, May 6, 2012 at 1:03 PM, Markus unive...@truemetal.org wrote: Am 06.05.2012 13
[asterisk-users] Problem with SendDTMF
Hello, I am having a problem with SendDTMF - it is not sending the numbers properly during the phone call.. I want the numbers key to to be pressed/sent automatically after 3 seconds during a phone call. I use software phone to test it... when I dialed 501, I cant hear anything for about 10 seconds (this is because of SendDTMF) and then I can hear the operator saying to enter the numbers but SendDTMF already did it?! Asterisk server are connected to voip.ms provider. I have spent many hours trying to get to work, how to fix this issue? See the configuration and debug log below: extensions.conf [test] exten = 501,1,Set(CALLERID(num)=004471XXX) exten = 501,n,Dial(SIP/+44797XX@voipms,30,M(sendnumber)t) exten = 501,n,Hangup() [macro-sendnumber] exten = s,1,Wait(3) exten = s,n,SendDTMF(www0w7w8w8wXwXwXwXwXwX) sip.conf == [general] context=default tcpbindaddr=0.0.0.0 dtmfmode = rfc2833 register = x:vxx...@london.voip.ms:5060 [test] type=peer secret=2xxx host=dynamic context=test [voipms] canreinvite=no host=london.voip.ms secret=xx type=peer username=135xxx ;your account disallow=all allow=gsm ; allow=g729 ; Uncomment if you support G729 fromuser=135xxx insecure=invite trustrpid=yes sendrpid=yes nat=yes dtmfmode=rfc2833 debug: = == Using SIP RTP CoS mark 5 -- Executing [501@test:1] Set(SIP/test-0026, CALLERID(num)=004471XX) in new stack -- Executing [501@test:2] Dial(SIP/test-0026, SIP/+4479XX@voipms,30,M(sendnumber)t) in new stack == Using SIP RTP CoS mark 5 -- Called +44797XX@voipms -- SIP/voipms-0027 is making progress passing it to SIP/test-0026 -- SIP/voipms-0027 answered SIP/test-0026 -- Executing [s@macro-sendnumber:1] Wait(SIP/voipms-0027, 3) in new stack -- Executing [s@macro-sendnumber:2] SendDTMF(SIP/voipms-0027, www0w7w8wXwXwXwXw4wXwXwX) in new stack Thanks! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
