Re: [asterisk-users] Asterisk system for church call center
We have a lot of clients who run small call centers based on Trixbox, and seem to be pretty happy with them. Have a look here: http://queuemetrics.com/manuals/QM_Trixbox-chunked/ Thanks l. 2010/3/31 Frank Church voi...@googlemail.com On 29 March 2010 21:46, Frank Church voi...@googlemail.com wrote: I have been asked by my church to recommend a VoIP system which can do the following. They do internet radio shows which are sometimes broadcast on radio. They are looking for a system which does the following for about 5 agents, exactly as they have described it. 1. Take incoming calls 2. Put them on hold if there is no one to handle the call immediately, or transfer them to an available agent 3. Take down their details, and number, (if this can be retrieved and saved from the caller id, thats better) 4. Get them to hold on after taking their details if they still want to hold 5. Call them back when the backlog is cleared up. I have a fairly good grasp of the hardware and programming part of Asterisk, having compiled it more than a few times and implemented A2Billing phone card and call shop system with it. But the type of software suited to the Call Center side is where my knowledge gap lies. I am looking for solutions based on the usual Asterisk distributions like AsteriskNow, trixbox, elastix etc, whether ready packaged or requiring additional customization. The matter of whether they will use soft phones, or regular phones with headsets is also something to consider. Soft phones with good GUI's may be preferred if more cost effective for them, although my personal preferences are with hard phones. Any recommendations - the ease of software for the end users is the main thing for me, and integration with the database for taking customers details is the main thing for me. One of the distributions with SugarCRM comes to mind here. Sorry for cross-posting, but ready made and commercially supported systems are not ruled out, if they come within their budget. Regards Frank Church After there response I will go with some of ready made Asterisk distributions, then consider to go for a commercial supported versions if they do not meet the churches needs. Thanks Frank -- Loway - home of QueueMetrics - http://queuemetrics.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Slightly more advanced dialling..
Hi, the system() part pointed me in the right direction.. Thanks, going to give it a test now.. Thanks! Andy On 29 March 2010 20:24, Zeeshan Zakaria zisha...@gmail.com wrote: Hi, I have done it a few times. Just posted a small blog about it with code. Check it at www.ilovetovoip.com/?p=322. I hope it'll help you. -- Zeeshan A Zakaria On 2010-03-29 11:07 AM, Philipp von Klitzing klitz...@pool.informatik.rwth-aachen.de wrote: Hi! I'm wondering if it is possible to ring X number of extensions simultaneously, and each answer... You might want to explain what you are trying to do. Dial() can handle this by using something like SIP/peer1SIP/peer2 The first one that answers wins. Look at the Dial option M to run a macro after the call has been answered. Also have a look at FollowMe() since it can do parallel calling. Or read up how to create a bunch of .call files using System() and a script. I can do a huntgroup-esque way of dialling, but I want all the dialled numbers to be picked up Do you mean to say: I want all dialed numbers to keep on ringing until they are answered, regardless if the initial callers has already been taken care of by the first extensions that reacted? In the Asterisk world, and usually in the PBX world in general, pick up has specific and different meaning (see *8 or app_pickup). Philipp -- _ -- Bandwidth and Colocation Pr... -- _ -- 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] Live Audio Streaming- From Aux interface-Online resource
Jonathan Addleman j...@redowl.ca wrote: nik600 wrote: I was trying to record a call usng Mixmonitor and then convert it using ffmpeg but the recording file is continuosly growing and ffmpeg ends the conversion before of the call completion. Here's my quick and easy eagi script: #!/bin/sh cat /dev/fd/3 | sox -t raw -r 8000 -w -s -c 1 - -t raw -r 44100 - vol 2| ffmpeg -f s16le -ar 44100 -ac 1 -i - -ab 32k -f mp3 - | ezstream -c /var/lib/asterisk/ices/stream.mp3.xml It just dumps the audio through sox, to increase the volume a bit, and convert the sample rate, then ffmpeg to encode the mp3, and then ezstream to send it to an icecast server. I could probably skip the sox step, and get ffmpeg to do those adjustments on its own, but for now, I know sox's command line better, so I used that. :) The dialplan is as simple as exten = meetme,n,MeetMe(confname,1qd) put all the members of the conversation in there, exten = mp3stream,n,EAGI(mp3stream.sh) and then put this in as well to start recording. What is the significance of /dev/fd/3 where does it come from? -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] a2billing wont pass the number
I think you have caller ID update set to Yes and A2Billing first asks you to: Enter your Caller ID number and then it asks you: Enter your destination number while you mistake both for destination number. Otherwise, I am confused by the title of your question that your caller id doesn't pass and that the message content is not related to it. -Bruce 2010/3/30 Juan E. Rodríguez jerdg...@gmail.com When you say 'a2billing' won't pass the number, you mean you are calling to an IVR or something like that. And when did you dial you destination number twice??? Saludos, Juan E. Rodríguez -Original Message- From: Nathanial Allan nathanial.al...@gmail.com Date: Tue, 30 Mar 2010 13:08:24 To: asterisk-users@lists.digium.com Subject: [asterisk-users] a2billing wont pass the number I am running into an issue with A2Billing. I will explain first of all that everything else works! the system is 90% complete its just this one small problem I am running into. So my problem is that when I place a call, 1. I dial my number that I want and A2Billing gets activated 2. it asks for my pin, upon successful entry of my pin A2Billing then 3. prompts me for my phone number then 4. The call goes out (and actually connects for the record) So I am entering my destination phone number twice which is not the worst thing that can happen, though it is a little annoying Any light that you can shine on this problem would be greatly appreciated as I have been working on it for too long now and I want to get a product! Thank You NallaN -- _ -- 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] Asterisk system for church call center
SugarCRM and the church. This sounds just like a business; one that doesn't like to call itself a business but employees tactics. I suggest providing them with a solid cisco system with 100s of thousands dollars in cost where they will have less money left to do bad things to world. Asterisk is too good for a church :) On Wed, Mar 31, 2010 at 3:32 PM, Lenz Emilitri lenz.lo...@gmail.com wrote: We have a lot of clients who run small call centers based on Trixbox, and seem to be pretty happy with them. Have a look here: http://queuemetrics.com/manuals/QM_Trixbox-chunked/ Thanks l. 2010/3/31 Frank Church voi...@googlemail.com On 29 March 2010 21:46, Frank Church voi...@googlemail.com wrote: I have been asked by my church to recommend a VoIP system which can do the following. They do internet radio shows which are sometimes broadcast on radio. They are looking for a system which does the following for about 5 agents, exactly as they have described it. 1. Take incoming calls 2. Put them on hold if there is no one to handle the call immediately, or transfer them to an available agent 3. Take down their details, and number, (if this can be retrieved and saved from the caller id, thats better) 4. Get them to hold on after taking their details if they still want to hold 5. Call them back when the backlog is cleared up. I have a fairly good grasp of the hardware and programming part of Asterisk, having compiled it more than a few times and implemented A2Billing phone card and call shop system with it. But the type of software suited to the Call Center side is where my knowledge gap lies. I am looking for solutions based on the usual Asterisk distributions like AsteriskNow, trixbox, elastix etc, whether ready packaged or requiring additional customization. The matter of whether they will use soft phones, or regular phones with headsets is also something to consider. Soft phones with good GUI's may be preferred if more cost effective for them, although my personal preferences are with hard phones. Any recommendations - the ease of software for the end users is the main thing for me, and integration with the database for taking customers details is the main thing for me. One of the distributions with SugarCRM comes to mind here. Sorry for cross-posting, but ready made and commercially supported systems are not ruled out, if they come within their budget. Regards Frank Church After there response I will go with some of ready made Asterisk distributions, then consider to go for a commercial supported versions if they do not meet the churches needs. Thanks Frank -- Loway - home of QueueMetrics - http://queuemetrics.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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 load balancing and failover
huu giang schrieb: Hi Zeeshan I know a solution using DRBD, Heartbeat and RedFone hardware to provide failover ability to Asterisk. If I have two Asterisk Servers, and each server has a TDM card and a PRI line connect to each card, how your solution can provide failover ability to Asterisk ? Do you need any other hardware? The calles to my IVR System don't just come from IP network (SIP) but can come from SS7 network. Well, if that case the SS7 Switch to which you are connected should be able to load balance the call to both of your servers. I guess you have two point codes for you servers? If one server goes down, the ss7 switch received the red alarms and stops to route calls to it. Once the server is up again it will get new calls. So, we only thing you have to worry about is to keep state information between the two servers consistent if people record messages or access databases. Regards, Tobias Thanks. --- On *Fri, 3/26/10, Zeeshan Zakaria /zisha...@gmail.com/* wrote: From: Zeeshan Zakaria zisha...@gmail.com Subject: Re: [asterisk-users] Asterisk load balancing and failover To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Friday, March 26, 2010, 1:51 AM About two years ago I setup two high availability solutions using DRBD and Heartbeat. The worked great and shutting down or unplugging one server stayed transparent for the callers, as IVRs stayed available. Having said this, it was not very straight forward to set it up, but not very difficut either. So Heartbeat and DRBD can be a good starting point for you. -- Zeeshan A Zakaria On 2010-03-26 4:40 AM, huu giang huugiang...@yahoo.com /mc/compose?to=huugiang...@yahoo.com wrote: Hi List, I'm finding a solution to provide failover and load balancing features to my IVR system. Anyone suggest me what is the best solution please?. what the hardware I should use ?. I heard about RedFone, but someone on the mail list said that it is not good because *TDMoE* module in asterisk is not so *stable* and TDMoE is stale. And It seems that RedFone doesn't not support load balancing ability (I can't find any document about this feature). Best Regards, Giang Huu. -- _ -- 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 -Inline Attachment Follows- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk load balancing and failover
Hi, Good to know this but I am not the poster of this question and not doing any load balancing. Zeeshan A Zakaria -- Sent from my Android phone with K-9 Mail. On 2010-03-31 7:33 AM, Tobias Wolf tobias.w...@evision.de wrote: huu giang schrieb: Hi Zeeshan I know a solution using DRBD, Heartbeat and RedFone hardware to provide failover... Well, if that case the SS7 Switch to which you are connected should be able to load balance the call to both of your servers. I guess you have two point codes for you servers? If one server goes down, the ss7 switch received the red alarms and stops to route calls to it. Once the server is up again it will get new calls. So, we only thing you have to worry about is to keep state information between the two servers consistent if people record messages or access databases. Regards, Tobias Thanks. --- On *Fri, 3/26/10, Zeeshan Zakaria /zisha...@gmail.com/* wrote: ... /mc/compose?to=huugiang...@yahoo.com wrote: Hi List, I'm finding a sol... -- _ -- Bandwidth and Colocation Pr... -- _ -- 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] Live Audio Streaming- From Aux interface-Online resource
Many thanks Jonathan! On Wed, Mar 31, 2010 at 10:29 AM, cov...@ccs.covici.com wrote: What is the significance of /dev/fd/3 where does it come from? I'ts the file descriptor 3 for the EAGI process, wich contains the audio. -- /*/ nik600 http://www.kumbe.it -- _ -- 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] Jitter Buffer and MeetMe.
Hello. I'm having Asterisk 1.6.0.x and trying to solve the issue concerning with a bad quality of voice for incoming SIP calls into the app_meetme. As I know, in my case of calls, jitter buffer is NOT executed on anyone channel. So, after reading Russell Bryant's post ( http://www.russellbryant.net/blog/2007/10/09/asterisk-jitterbuffer-support-for-applications/) I added following scheme in dialplan: [some-context] exten = 123,1,Dial(Local/124 at some-context/nj) exten = 124,1,MeetMe(some-room,dM) So, the problem with voice quality was completely solved, BUT some customers have informed me about big latency. It's really hard to make dialogue with current latency. And there are some questions: 1. Where can I find the best practice to solve the issue with JB and applications (MeetMe)? 2. Is it possible to adjust (reduce) generic JB in chan_local and for Local/.../nj construction? BR, Alexey -- _ -- 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] Live Audio Streaming- From Aux interface-Online resource
OK, I see, but what I would really like to do is the opposite -- stream an internet stream into a call or a meetme conference -- what would be the best way on how to do that? nik600 nik...@gmail.com wrote: Many thanks Jonathan! On Wed, Mar 31, 2010 at 10:29 AM, cov...@ccs.covici.com wrote: What is the significance of /dev/fd/3 where does it come from? I'ts the file descriptor 3 for the EAGI process, wich contains the audio. -- /*/ nik600 http://www.kumbe.it -- _ -- 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 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Live Audio Streaming- From Aux interface-Online resource
On Wed, Mar 31, 2010 at 2:17 PM, cov...@ccs.covici.com wrote: OK, I see, but what I would really like to do is the opposite -- stream an internet stream into a call or a meetme conference -- what would be the best way on how to do that? And (hijacking thread with related question) I'd like to stream from an incoming leg of a SIP channel to the Internet. Any suggestions on that? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Unable to login to voicemail with Ekiga
Hello, Asterisk 1.4.26.2, FreeBSD 8.0-RELEASE We have a very simple setup, using SIP softphones and a simple diaplan as follows in the examples below. When I dial the 700 extension it asks me for the extension and password, and it always says login incorrect. The mail system send the email ok and Ekiga shows that I have vaoicemail, so the only thing that is failing is the actual login to the mailbox. I have searched many threads, and most if not all, talk abot the dtmf setiings, but both Ekiga and Asterisk are configured for rfc2833. Here is what I get in the console: [Mar 30 10:30:23] WARNING[1790]: app_voicemail.c:7236 vm_authenticate: Couldn't read username Thanks beforehand! Alejandro Imass sip.conf [101] username=101 type=friend secret=xx qualify=yes nat=no host=dynamic canreinvite=no context=home mailbox=...@home dtmfmode=rfc2833 extensions.conf [home] ...snip... ;internal sip extensions exten = 101,1,Dial(SIP/101,15) exten = 101,2,Voicemail(1...@home) ...snip... ;voice mail exten = 700,1,VoiceMailMain() ...snip... voicemail.conf [home] 101 = ,User Name,u...@domain -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unable to login to voicemail with Ekiga
The message Couldn't read user name means it is not receiving the DTMF. Do you have an IVR to verify that your system is receiving the DTMF? If not, setup one, call into it and send Dtmf to it and see if it responds at all. If it doesn't, somewhere DTMF settings need to be adjusted. Zeeshan A Zakaria -- Sent from my Android phone with K-9 Mail. On 2010-03-31 9:15 AM, Alejandro Imass a...@p2ee.org wrote: Hello, Asterisk 1.4.26.2, FreeBSD 8.0-RELEASE We have a very simple setup, using SIP softphones and a simple diaplan as follows in the examples below. When I dial the 700 extension it asks me for the extension and password, and it always says login incorrect. The mail system send the email ok and Ekiga shows that I have vaoicemail, so the only thing that is failing is the actual login to the mailbox. I have searched many threads, and most if not all, talk abot the dtmf setiings, but both Ekiga and Asterisk are configured for rfc2833. Here is what I get in the console: [Mar 30 10:30:23] WARNING[1790]: app_voicemail.c:7236 vm_authenticate: Couldn't read username Thanks beforehand! Alejandro Imass sip.conf [101] username=101 type=friend secret=xx qualify=yes nat=no host=dynamic canreinvite=no context=home mailbox=...@home dtmfmode=rfc2833 extensions.conf [home] ...snip... ;internal sip extensions exten = 101,1,Dial(SIP/101,15) exten = 101,2,Voicemail(1...@home) ...snip... ;voice mail exten = 700,1,VoiceMailMain() ...snip... voicemail.conf [home] 101 = ,User Name,u...@domain -- _ -- 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] convert from wav or mp3 to gsm
I use this all the time and am very pleased with the results... sox ORIG_FILE.WAV -r 8000 -c 1 OUTPUT_FILE.gsm resample -ql From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Tuesday, March 30, 2010 3:29 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] convert from wav or mp3 to gsm AIR, * uses wav and gsm with no trouble. Mpg123 plays mp3 format files. You can use LAME and SOX to change files between these formats. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of salaheddine elharit Sent: Tuesday, March 30, 2010 3:17 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] convert from wav or mp3 to gsm Hello All do you have ant software in order to change the format from mp3 or wav to gsm in order to using it in asterisk file thank you so much for your help and support Best Regards, salah -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unable to login to voicemail with Ekiga
On Wed, Mar 31, 2010 at 9:23 AM, Zeeshan Zakaria zisha...@gmail.com wrote: The message Couldn't read user name means it is not receiving the DTMF. Do you have an IVR to verify that your system is receiving the DTMF? If not, setup one, call into it and send Dtmf to it and see if it responds at all. If it doesn't, somewhere DTMF settings need to be adjusted. The IVR works fine, and we use it everyday. That's why it seemed to me that it could not be a stmf problem. Any other ideas? Zeeshan A Zakaria -- Sent from my Android phone with K-9 Mail. On 2010-03-31 9:15 AM, Alejandro Imass a...@p2ee.org wrote: Hello, Asterisk 1.4.26.2, FreeBSD 8.0-RELEASE We have a very simple setup, using SIP softphones and a simple diaplan as follows in the examples below. When I dial the 700 extension it asks me for the extension and password, and it always says login incorrect. The mail system send the email ok and Ekiga shows that I have vaoicemail, so the only thing that is failing is the actual login to the mailbox. I have searched many threads, and most if not all, talk abot the dtmf setiings, but both Ekiga and Asterisk are configured for rfc2833. Here is what I get in the console: [Mar 30 10:30:23] WARNING[1790]: app_voicemail.c:7236 vm_authenticate: Couldn't read username Thanks beforehand! Alejandro Imass sip.conf [101] username=101 type=friend secret=xx qualify=yes nat=no host=dynamic canreinvite=no context=home mailbox=...@home dtmfmode=rfc2833 extensions.conf [home] ...snip... ;internal sip extensions exten = 101,1,Dial(SIP/101,15) exten = 101,2,Voicemail(1...@home) ...snip... ;voice mail exten = 700,1,VoiceMailMain() ...snip... voicemail.conf [home] 101 = ,User Name,u...@domain -- _ -- 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] convert from wav or mp3 to gsm
On Tue, Mar 30, 2010 at 4:16 PM, salaheddine elharit salah.elharit...@gmail.com wrote: Hello All do you have ant software in order to change the format from mp3 or wav to gsm in order to using it in asterisk file thank you so much for your help and support Best Regards, salah If you use a 1.6 series asterisk, you can build mp3 channel support, right? make menuconfig on the source tree, and add it. Or is it in extras? -- _ -- 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] 24 FXS Port Voip Gateway and Asterisk
Another option is to tie in a legacy 2-wire PBX with Asterisk instead of going pure analog This allows you to reuse your single-pair infrastructure, while achieving MOST of the functionality of a pure-ip endpoint deployment with only a very moderate incremental cost over a pure-analog deployment. The upside is that you can get things that the analog devices can't give you, such as network time, paging, speakerphones, talkback and lots of hardware buttons for 1-click access to limitless asterisk features. Any legacy PBX is also going to be compatible with your analog devices such as mailing machines (modems) and Faxes (if you still use them) via their 'synchronous' (not-packetized) ATA's or analog 'ports'. Consider how cheap SOLID STATE Norstar equipment is for example. A few hundred bucks for a perfectly good PRI-equipped decommissioned system that is DISKLESS FANLESS (read high availability) along high-quality speakerphone endpoints for virtually nothing (on eBay), or dirt-cheap 'refurb' equipment that is tested warranted. Don't get me wrong, I prefer the power flexibility of a POE managed switch IP endpoints, and without a doubt a pure-analog system is far simpler, but if you have cost constraints or physical constraints, and want more functionality that pure-analog can give you, an asterisk-equipped legacy PBX is a powerful, flexible option that should not be overlooked. -Karl - Original Message - From: Joseph Begumisa To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Tuesday, March 30, 2010 4:34 PM Subject: Re: [asterisk-users] 24 FXS Port Voip Gateway and Asterisk And not to mention the need for power over ethernet switches to avoid having many power adpaters lying all over. Don't get me wrong, I'm for IP Phones, however, in this specific scenario that I have, getting an FXS to SIP gateway with 24 ports makes more sense. Thanks for all the pointers. Best Regards, Joseph On Tue, Mar 30, 2010 at 8:39 AM, Andrew Latham lath...@gmail.com wrote: And to add to this, analog is useful for its distance when running wall phones in a large warehouse setting... ~ Andrew lathama Latham lath...@gmail.com * Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software * Learn more about Linux http://en.wikipedia.org/wiki/Linux * Learn more about Tux http://en.wikipedia.org/wiki/Tux On Tue, Mar 30, 2010 at 11:29 AM, Darrick Hartman dhart...@djhsolutions.com wrote: Sometimes you need to look at the cost to pull new wire too, not just the cost of the phones. There are a few cases where the channel banks + analog phones make sense, especially when the analog devices are already there. Sent from my BlackBerry® wireless device from U.S. Cellular -Original Message- From: hin lee hi...@yahoo.com Date: Tue, 30 Mar 2010 08:25:19 To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: Re: [asterisk-users] 24 FXS Port Voip Gateway and Asterisk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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
Re: [asterisk-users] convert from wav or mp3 to gsm
A (hopefully) helpful addition to this reply; ORIG_FILE.WAV is a wav49 file. If the name is orig_file.wav, it is a regular wav file and the sox command would generate (IMO) a better output like this: sox orig_file.wav.WAV -r 8000 -v 10 -c 1 OUTPUT_FILE.gsm resample -ql _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Robert Grignon Sent: Wednesday, March 31, 2010 8:50 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] convert from wav or mp3 to gsm I use this all the time and am very pleased with the results... sox ORIG_FILE.WAV -r 8000 -c 1 OUTPUT_FILE.gsm resample -ql _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Tuesday, March 30, 2010 3:29 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] convert from wav or mp3 to gsm AIR, * uses wav and gsm with no trouble. Mpg123 plays mp3 format files. You can use LAME and SOX to change files between these formats. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of salaheddine elharit Sent: Tuesday, March 30, 2010 3:17 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] convert from wav or mp3 to gsm Hello All do you have ant software in order to change the format from mp3 or wav to gsm in order to using it in asterisk file thank you so much for your help and support Best Regards, salah -- _ -- 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] convert from wav or mp3 to gsm
Hi, Oki, thank you so much for this solution i really appreciate it Regards, Salah 2010/3/31 Danny Nicholas da...@debsinc.com A (hopefully) helpful addition to this reply; ORIG_FILE.WAV is a wav49 file. If the name is orig_file.wav, it is a regular wav file and the sox command would generate (IMO) a better output like this: sox orig_file.wav.WAV -r 8000 –v 10 -c 1 OUTPUT_FILE.gsm resample -ql -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Robert Grignon *Sent:* Wednesday, March 31, 2010 8:50 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] convert from wav or mp3 to gsm I use this all the time and am very pleased with the results... sox ORIG_FILE.WAV -r 8000 -c 1 OUTPUT_FILE.gsm resample -ql -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Danny Nicholas *Sent:* Tuesday, March 30, 2010 3:29 PM *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' *Subject:* Re: [asterisk-users] convert from wav or mp3 to gsm AIR, * uses wav and gsm with no trouble. Mpg123 plays mp3 format files. You can use LAME and SOX to change files between these formats. -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *salaheddine elharit *Sent:* Tuesday, March 30, 2010 3:17 PM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] convert from wav or mp3 to gsm Hello All do you have ant software in order to change the format from mp3 or wav to gsm in order to using it in asterisk file thank you so much for your help and support Best Regards, salah -- _ -- 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] Dropped Calls
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of JR Richardson Sent: Tuesday, March 30, 2010 6:55 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Dropped Calls I've written about this issue several times, but have not yet found any solution to it. I am using asterisk 1.4.21.2 and zaptel 1.4.12. Phones are primarily Snom 300's but I also have a couple of headset phones connected to Grandstream HT286 SIP adapters. I have 8 offices, each has it's own asterisk server all running the same versions of asterisk and Zaptel. Only difference is that one office uses a Digium TDM 8-port card and the other branches use 4-port Rhino cards with only 2 ports in use. What happens is that periodically we will be in a call and the call will just drop. It's usually within the first couple of minutes of the call. The calls can be either incoming or outgoing. The phenomenon affects both the Snoms and the Grandstreams. Along with the dropped call issue, we periodically have a problem where a person we call or a person that calls in cannot hear the person in the our office, but the person in our office can hear the remote person fine. All of the phones are on the same physical network as the asterisk server. There is no NAT, no Firewall, VLAN, etc. between the phones and the server. I have tried running sip debugs on the calls, but on the off chance that my logs catch either a drop or a one-way audio, the sip debug looks like just a normal call. Is there any setting that might cause both one-way audio and dropped calls? Thanks, Brent Davidson Join the club. I've experienced the same with various strains on 1.4.x above 1.4.21.1 (not an issue with this one that I have seen). This issue is truly random and debugging reveals nothing. I run an all SIP environment with same results. My solution was to downgrade to another version or switch to 1.2 or 1.6 depending on what features I need for the system. Sorry I couldn't be of any help, but I feel your frustration. JR -- JR Richardson Engineering for the Masses Is there a chance that you are using Realtime at all? I am just curious because I was having problems with dropped calls as well and just discovered that it appears to be related to the database server. If for some reason on the database server there is a table lock (which I am investigating why) asterisk drops any PRI calls and SIP calls. Everything looked normal and the error messages never once suggest a problem with the database server or Realtime. I was looking everywhere else but at the Realtime until I stumbled across it. While doing some backups with FLUSH READ LOCKS to a slave machine, which I changed asterisk to use a few months back, I had dropped calls occur. I later confirmed that asterisk seems to hang / freeze during that period but once the database server releases the locks, asterisk continues to function without any problems. This started to occur when we had an increase in call volume and an increase in load on the db server. I was using Realtime for extensions, sip peers and CDR. I had turned off using realtime for CDR (which we don't really use anyway) and started to use a slave server instead of the master when performing some maintenance on the master db server. I left it that way since I was just using it for extensions and sip peers and that had cleared it up over the last few months until I ran my backup. Not sure that helps but it is worth a shot in mentioning to you. Regards, Michael Young (elguero) -- _ -- 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] Reset personal voicemail settings
Hi list, can anyone tell me how to reset/delete all modifications (personal greeting message, personal name, ...) I made in my voicemail? I just want to get the default automatic computer messages back. thank you! greets felix -- _ -- 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] Dropped Calls
On 3/31/2010 10:38 AM, Michael L. Young wrote: Is there a chance that you are using Realtime at all? I am just curious because I was having problems with dropped calls as well and just discovered that it appears to be related to the database server. If for some reason on the database server there is a table lock (which I am investigating why) asterisk drops any PRI calls and SIP calls. Everything looked normal and the error messages never once suggest a problem with the database server or Realtime. I was looking everywhere else but at the Realtime until I stumbled across it. While doing some backups with FLUSH READ LOCKS to a slave machine, which I changed asterisk to use a few months back, I had dropped calls occur. I later confirmed that asterisk seems to hang / freeze during that period but once the database server releases the locks, asterisk continues to function without any problems. This started to occur when we had an increase in call volume and an increase in load on the db server. I was using Realtime for extensions, sip peers and CDR. I had turned off using realtime for CDR (which we don't really use anyway) and started to use a slave server instead of the master when performing some maintenance on the master db server. I left it that way since I was just using it for extensions and sip peers and that had cleared it up over the last few months until I ran my backup. Not sure that helps but it is worth a shot in mentioning to you. Regards, Michael Young (elguero) In my case, no. All extensions are hard-coded. We only have a handful of phones that don't change. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unable to login to voicemail with Ekiga
On Wed, Mar 31, 2010 at 9:23 AM, Zeeshan Zakaria zisha...@gmail.com wrote: The message Couldn't read user name means it is not receiving the DTMF. Do you have an IVR to verify that your system is receiving the DTMF? If not, setup one, call into it and send Dtmf to it and see if it responds at all. If it doesn't, somewhere DTMF settings need to be adjusted. Zeeshan A Zakaria Is this the right list? or is this for final users? -- Sent from my Android phone with K-9 Mail. -- _ -- 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] Live Audio Streaming- From Aux interface-Online resource
Hi! And (hijacking thread with related question) I'd like to stream from an incoming leg of a SIP channel to the Internet. Any suggestions on that? Start here: http://www.voip-info.org/wiki/view/Asterisk+cmd+ices Philipp -- _ -- 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] Live Audio Streaming- From Aux interface-Online resource
On Wed, Mar 31, 2010 at 6:27 PM, Philipp von Klitzing klitz...@pool.informatik.rwth-aachen.de wrote: Start here: http://www.voip-info.org/wiki/view/Asterisk+cmd+ices Thanks, Philipp -- _ -- 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] Dropped Calls
Just to get a 100% correct response to last question, are you using the flat CDR or mysql/some other DB? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Brent Davidson Sent: Wednesday, March 31, 2010 10:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Dropped Calls On 3/31/2010 10:38 AM, Michael L. Young wrote: Is there a chance that you are using Realtime at all? I am just curious because I was having problems with dropped calls as well and just discovered that it appears to be related to the database server. If for some reason on the database server there is a table lock (which I am investigating why) asterisk drops any PRI calls and SIP calls. Everything looked normal and the error messages never once suggest a problem with the database server or Realtime. I was looking everywhere else but at the Realtime until I stumbled across it. While doing some backups with FLUSH READ LOCKS to a slave machine, which I changed asterisk to use a few months back, I had dropped calls occur. I later confirmed that asterisk seems to hang / freeze during that period but once the database server releases the locks, asterisk continues to function without any problems. This started to occur when we had an increase in call volume and an increase in load on the db server. I was using Realtime for extensions, sip peers and CDR. I had turned off using realtime for CDR (which we don't really use anyway) and started to use a slave server instead of the master when performing some maintenance on the master db server. I left it that way since I was just using it for extensions and sip peers and that had cleared it up over the last few months until I ran my backup. Not sure that helps but it is worth a shot in mentioning to you. Regards, Michael Young (elguero) In my case, no. All extensions are hard-coded. We only have a handful of phones that don't change. -- _ -- 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] Dropped Calls
Hi! I am just curious because I was having problems with dropped calls as well Maybe rtptimeout in sip.conf is involved (and not behaving as expected)? All extensions are hard-coded. We only have a handful of phones that don't change. This last sentence is a wounderful example of a sentence that can be interpreted in two, and very opposite, ways. :-) Philipp -- _ -- 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] meetme() and dahdi_dummy on an embedded system
Hi, Vinicius, did you actually solve the choppy audio issue by compiling Gordon's kernel? I have the same problem on the exact same Alix platform (using kernel 2.6.31, though). regards, Darko ps. Sorry everyone if this mail does not get threaded right. I've just joined the mailing list and not sure how to reply to an existing thread I found on the web. -- _ -- 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] Safe_asterisk doesn't exists???
Thanks Andrew, I have some doubts regarding this issue, firstly, i'm using Asterisk 1.4.21 not 1.6 like the issue you showed to me ( https://issues.asterisk.org/view.php?id=16887) other thing is that i have many other asterisk servers working good and i never made this change By the way i'm using Centos and i can't find the line: start-stop-daemon --start --oknodo --background --exec $DAEMON -- $ASTARGS Is this a bug from Asterisk 1.6 only? Thanks in advance for your help my friend -- Message: 4 Date: Wed, 24 Mar 2010 13:59:04 -0400 From: Andrew Latham lath...@gmail.com Subject: Re: [asterisk-users] Safe_asterisk doesn't exists??? To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: c39c115d1003241059q5852f6a2y7c1e81b8ef1fa...@mail.gmail.com Content-Type: text/plain; charset=UTF-8 https://issues.asterisk.org/view.php?id=16887 do a make update ~ Andrew lathama Latham lath...@gmail.com * Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software * Learn more about Linux http://en.wikipedia.org/wiki/Linux * Learn more about Tux http://en.wikipedia.org/wiki/Tux On Tue, Mar 23, 2010 at 7:16 PM, Danny Dias ing.diasda...@gmail.com wrote: Hello my friends, I'm very worry about a problem i'm having...my asterisk got freez some times, every 5 or 6 days with NO trace in /var/log/asterisk/messages What i want to know is if safe_asterisk has something to be with this? This is what i have on my server: [r...@mypbx ~]# ps -A | grep asterisk ?9118 ? ? ? ? ?00:01:30 asterisk [r...@dreampbx ~]# ps aux | grep asterisk root ? ? ?9118 ?0.1 ?0.3 29668 12520 ? ? ? ? Sl ? Mar22 ? 1:30 /usr/sbin/asterisk -f -vvvg -c root ? ? 12096 ?0.0 ?0.0 ?4140 ?640 pts/1 ? ?S+ ? 18:40 ? 0:00 grep asterisk I have another asterisk servers working and the commands above always shows safe _asterisk as a process... This safe_asterisk could be the cause of my problems? how does it works? how can i activate it? Thanks in advance for your valuable help! DD -- _ -- 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] Multicast Paging
I know this may be a bit off topic...
I'm trying to play a pre-recorded message to a group of Aastra phones using
multicast paging. I can page phone to phone without issue, but sending from one
of my servers to the phones results in garbled audio. Anyone else been able to
make this work without problem? My VLC command line is below.
cvlc -v emergency-test2.wav --norm-max-level=5 --sout
#transcode{acodec=ulaw,ab=64,channels=1,samplerate=8000}:rtp{dst=239.0.1.20,port-audio=16000,proto=udp}
-Jon
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Re: [asterisk-users] DAHDI 2.2.1, Asterisk 1.6.2.6 - Channel unacceptable (6)
Hi, it was some configuration error. I droped the old config and started with the sample file to build a new one. Therefore I do not know which parameter in chan_dahdi.conf caused the problem. Anyway, now it is working. The only remaining problem is that no caller ID is available for incoming calls. -- Stefan Tichy ( asterisk2 at pi4tel dot de ) -- _ -- 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] Live Audio Streaming- From Auxinterface-Online resource
According to this link http://www.voip-info.org/wiki/view/Asterisk+cmd+MeetMe , you could pipe the stream into the conference using an AGI script. I haven't actually tried it. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of cov...@ccs.covici.com Sent: Wednesday, March 31, 2010 7:18 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Live Audio Streaming- From Auxinterface-Online resource OK, I see, but what I would really like to do is the opposite -- stream an internet stream into a call or a meetme conference -- what would be the best way on how to do that? nik600 nik...@gmail.com wrote: Many thanks Jonathan! On Wed, Mar 31, 2010 at 10:29 AM, cov...@ccs.covici.com wrote: What is the significance of /dev/fd/3 where does it come from? I'ts the file descriptor 3 for the EAGI process, wich contains the audio. -- /*/ nik600 http://www.kumbe.it -- _ -- 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 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] Multicast Paging
Jonathan C. Bailey wrote:
I know this may be a bit off topic...
I'm trying to play a pre-recorded message to a group of Aastra phones using
multicast paging. I can page phone to phone without issue, but sending from
one of my servers to the phones results in garbled audio. Anyone else been
able to make this work without problem? My VLC command line is below.
cvlc -v emergency-test2.wav --norm-max-level=5 --sout
#transcode{acodec=ulaw,ab=64,channels=1,samplerate=8000}:rtp{dst=239.0.1.20,port-audio=16000,proto=udp}
-Jon
Why not use the built in multicast paging system? :)
https://issues.asterisk.org/view.php?id=11797
It appears to exist in 1.6.2.
Leif.
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[asterisk-users] No audio when calling via PSTN, before remote answers (with polarity reversal)
Hi! I want to get audio from the PSTN before the call is answered so I don't miss when the called phone is busy or if there is some error (like the phone is unavailable or is wrong, etc) and hear the ringing from my telco. I have polarity reversal in my telco for incoming and outgoing calls. If I set answeronpolarityswitch=yes then I get no audio until the call is answered. If I set it to no it works fine sometimes, but other times when the call gets answered, asterisk detects the polarity reversal as a hangup and hangs up the call. I need to have hanguponpolarityswitch set to yes for detecting hang ups in incoming calls. (it was a nightmare before this) Any ideas? I had this working in previous versions of asterisk but didn't find what was the change that caused this behaviour, and I need to use a recent version of asterisk for some changes in dtmf caller id detection that aren't in my distro yet (debian). Thanks in advance. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Upcoming Asterisk 1.6.0 and 1.6.1 Maintenance Changes
Maintenance of Asterisk 1.6.0 and 1.6.1 will move to security fixes only in approximately one month. There are bug fix releases scheduled to be released during the first half of May for both versions. After those releases, Asterisk 1.6.0 and 1.6.1 will only receive security fixes. The Asterisk development team recommends that all users of Asterisk 1.6.0 and 1.6.1 series move to the 1.6.2 series for continued bug fix support. For more information on the maintenance schedule for Asterisk releases, please see the following page: http://www.asterisk.org/asterisk-versions Note that the maintenance schedule for Asterisk 1.4 and 1.6.2 will likely be extended, pending the final determination of the release schedule for Asterisk 1.8. Thank you for your continued support of Asterisk! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to run Music while looking for the caller in Database
Hi, We are using Asterisk and PERL. We have all the call logic in PERL. We are trying to identify the caller using the CID in the Database. As the Database lookup is taking more time (15 seconds), we want to play some tune while the caller is waiting. How can we do that? Any ideas will be greatly appreciated. - Bharath -- _ -- 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] How to run Music while looking for the caller in Database
Bharath B. Reddy Bynagari wrote: Hi, We are using Asterisk and PERL. We have all the call logic in PERL. We are trying to identify the caller using the CID in the Database. As the Database lookup is taking more time (15 seconds), we want to play some tune while the caller is waiting. How can we do that? Any ideas will be greatly appreciated. - Bharath Assuming that you are using a perl AGI script, you can use the AGI command SET MUSIC ON to play music on the channel. You can stop the music by using SET MUSIC OFF. For a bit more information, you can run the CLI command agi show set music Mark Michelson -- _ -- 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] Multicast Paging
Hi!
I'm trying to play a pre-recorded message to a group of Aastra phones
[...]
cvlc -v emergency-test2.wav --norm-max-level=5 --sout
#transcode{acodec=ulaw,ab=64,channels=1,samplerate=8000}:rtp{dst=239.0.1.
20,port-audio=16000,proto=udp}
Look at the very bottom of this (snom multicast):
http://www.voip-info.org/wiki/view/Asterisk+cmd+Page
I think I ran into this as well and then turned to feeding ulaw to vlc
instead of wav. And MAST is a good alternative to try.
Cheers, Philipp
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Re: [asterisk-users] How to run Music while looking for the caller in Database
On Wed, 31 Mar 2010, Bharath B. Reddy Bynagari wrote: We are using Asterisk and PERL. We have all the call logic in PERL. We are trying to identify the caller using the CID in the Database. As the Database lookup is taking more time (15 seconds), Fix the database! Anything else is a band-aid. What DBM? How many rows? What indexes? How long does it take to execute your query from a shell? we want to play some tune while the caller is waiting. You can create a separate thread in your AGI to play the Please wait while... while you do your database stuff. I did this several years ago with an AGI written in C to process a credit card authorization request and response -- but I was only trying to hide a task that took a second or two. The customer experience was that the authorization was instantaneous. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Reset personal voicemail settings
Felix Tiefenthaler wrote: Hi list, can anyone tell me how to reset/delete all modifications (personal greeting message, personal name, ...) I made in my voicemail? I just want to get the default automatic computer messages back. thank you! greets felix If you are storing voicemail on the file system, then you can just go to /var/spool/asterisk/voicemail/context/mailbox/ and delete the items in there that you want to. The INBOX and Old folders contain new and old messages. Anything else in there will be greetings and other similar recordings. Mark Michelson -- _ -- 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] Multicast Paging
I think I may have to do that.. I'm beginning to think my idea with VLC just
won't work. BTW, we're running 1.4.28 (but so far there seems to be a backport).
- Original Message -
From: Leif Madsen leif.mad...@asteriskdocs.org
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, March 31, 2010 2:42:14 PM GMT -06:00 US/Canada Central
Subject: Re: [asterisk-users] Multicast Paging
Jonathan C. Bailey wrote:
I know this may be a bit off topic...
I'm trying to play a pre-recorded message to a group of Aastra phones using
multicast paging. I can page phone to phone without issue, but sending from
one of my servers to the phones results in garbled audio. Anyone else been
able to make this work without problem? My VLC command line is below.
cvlc -v emergency-test2.wav --norm-max-level=5 --sout
#transcode{acodec=ulaw,ab=64,channels=1,samplerate=8000}:rtp{dst=239.0.1.20,port-audio=16000,proto=udp}
-Jon
Why not use the built in multicast paging system? :)
https://issues.asterisk.org/view.php?id=11797
It appears to exist in 1.6.2.
Leif.
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[asterisk-users] Necessary hardware
Hi there! just a quick question: what would you recommend to get to connect an asterisk box to the analog phoneline? I have two linksys spa2102 and a sap9000 but as far as I know I need something else to connect the asterisk box to the analog phoneline. I just have two analog phone lines, so getting one device to plug both is fine, but two devices to connect one by one would be good too. The cheaper the better this time. Thanks in advance. -- Kosa - Un mundo mejor es posible - -- _ -- 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] Dropped Calls
On 3/31/2010 12:06 PM, Danny Nicholas wrote: Just to get a 100% correct response to last question, are you using the flat CDR or mysql/some other DB? All sip clients/peers are defined in sip.conf, dial-plan is entirely in extensions.ael. We have one office that uses an Asterisk native database call in the dialplan for the operator extension to see which extension is currently handling operator calls, but other than that there is no no DB used on any of the other systems. -Brent -- _ -- 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] Dropped Calls
On 3/31/2010 12:16 PM, Philipp von Klitzing wrote: Maybe rtptimeout in sip.conf is involved (and not behaving as expected)? I was suspecting something with either rtptimeout or sip registration timeout, but I'm not sure what. -- _ -- 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 load balancing and failover
Do you mean that SS7 switch is a MSC and do all MSC support load balancing without any hardware between it and my Server. Sorry for my English, what do you mean two point codes for my servers ?. I have at least two servers. --- On Wed, 3/31/10, Tobias Wolf tobias.w...@evision.de wrote: From: Tobias Wolf tobias.w...@evision.de Subject: Re: [asterisk-users] Asterisk load balancing and failover To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Wednesday, March 31, 2010, 4:27 AM huu giang schrieb: Hi Zeeshan I know a solution using DRBD, Heartbeat and RedFone hardware to provide failover ability to Asterisk. If I have two Asterisk Servers, and each server has a TDM card and a PRI line connect to each card, how your solution can provide failover ability to Asterisk ? Do you need any other hardware? The calles to my IVR System don't just come from IP network (SIP) but can come from SS7 network. Well, if that case the SS7 Switch to which you are connected should be able to load balance the call to both of your servers. I guess you have two point codes for you servers? If one server goes down, the ss7 switch received the red alarms and stops to route calls to it. Once the server is up again it will get new calls. So, we only thing you have to worry about is to keep state information between the two servers consistent if people record messages or access databases. Regards, Tobias Thanks. --- On *Fri, 3/26/10, Zeeshan Zakaria /zisha...@gmail.com/* wrote: From: Zeeshan Zakaria zisha...@gmail.com Subject: Re: [asterisk-users] Asterisk load balancing and failover To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Friday, March 26, 2010, 1:51 AM About two years ago I setup two high availability solutions using DRBD and Heartbeat. The worked great and shutting down or unplugging one server stayed transparent for the callers, as IVRs stayed available. Having said this, it was not very straight forward to set it up, but not very difficut either. So Heartbeat and DRBD can be a good starting point for you. -- Zeeshan A Zakaria On 2010-03-26 4:40 AM, huu giang huugiang...@yahoo.com /mc/compose?to=huugiang...@yahoo.com wrote: Hi List, I'm finding a solution to provide failover and load balancing features to my IVR system. Anyone suggest me what is the best solution please?. what the hardware I should use ?. I heard about RedFone, but someone on the mail list said that it is not good because *TDMoE* module in asterisk is not so *stable* and TDMoE is stale. And It seems that RedFone doesn't not support load balancing ability (I can't find any document about this feature). Best Regards, Giang Huu. -- _ -- 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 -Inline Attachment Follows- -- _ -- 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
