[asterisk-users] Different IP addresss for SIP and RTP
Hello, is it possible to set an IP address for RTP different than the one used for SIP? I want to use asterisk behind a sip proxy (opensips), but I was thinking if I could avoid having to run rtpproxy on the sip proxy server and let asterisk itself take care of it. So that: Asterisk SIP address : local ip address Asterisk RTP address : global ip address regards, takeshi -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8.4 Now Available
Paul, I have kind of a related question. asterisk-1.8.4-summary.txt does not always properly link specific patches to issues. For example, revision 307509 is associated with issue 18542, and it is not reflected in the summary. There may be more like this. I tried to report this inconsistency timely, issue #18933, but it is still listed as new What is the right way of reporting documentation issues? Thank you, Vladimir On 5/11/2011 9:46 AM, Paul Belanger wrote: On 11-05-11 10:29 AM, Jeremy Kister wrote: I'm a bit confused about this release (and previous releases on the 1.8 track) so please bare with me. I viewed the ChangeLog, but I don't see any of the 'sample issues' listed. why is that ? I would expect to see the 'sample issues' listed after 1.8.4-rc3. Also, is there a reason/procedural error that patches such as: https://issues.asterisk.org/view.php?id=18382 https://issues.asterisk.org/view.php?id=18742 didnt make it into this 1.8.4 release ? Correct, they will appear in 1.8.5-rc1 forward. When -rc1 is created, it will be tagged from the HEAD of branches/1.8.. If 1.8.5-rc2 is create, it is because of an issue / bug was found in 1.8.5-rc1, and will include that fix only. If a new issue is reported after 1.8.5-rc1 and fixed in branches/1.8, it will not be added into 1.8.5 release, but will wait until 1.8.6-rc1. -- _ -- 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] With what options is asterisk compiled in rpm's
On 5/11/2011 10:15 AM, Danny Nicholas wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of isr...@gmail.com Sent: Wednesday, May 11, 2011 10:07 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] With what options is asterisk compiled in rpm's Hi, I'm trying to add modules compiled from source into a rpm install of asterisk (from digium) on centos and asterisk complains that its not compiled with same options so it won't load it I know I could install the entire thing from source but for other reasons I would like to keep the main things installed from rpm and install whatever else I need from source (or roll my own rpm for those) Thanks, Israel [Danny Nicholas] This might work for you - download the source and do ./configure --libdir=/fromsrclib --bindir=/fromsrcbin Once you've done the make menuselect, make and make install, you should be able to copy the module(s) from /fromsrclib to /usr/lib/asterisk/modules Danny, Can you please clarify. Are the /fromsrclib and the /fromsrcbin actual directories or keywords? If these are directories should they be created empty or populated with something? Thank you, Vladimir -- _ -- 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] When someone helps you, at least let them know if the problem is resolved or not
[This is my last post in this thread - as I really CBA anymore!] Wow! You really don't see it do you? Fair enough. I thought you were just playing along with my 'ego baiting' game - but it seems I hit the mother load of all ego's here. Apologies to all 'watchers' - but this was intended as a bit of fun - and I thought Steve was just playing along. Seems I was wrong. As a parting gesture though - I'll give you an example from your first post in here (which reads more like a CV/resume than a post! [in fact, they all do]): I was the number 1 poster on this list a couple of years ago I don't really do job searches, I am usually offered a job or project and approached by the client. My last trip was to Iraq, but I have been to Senegal, Sierra Leone, Guinea, Ghana, Liberia to help rebuild the infrastructure for USAID. For the Dept of State, I set up... For DoD/Dos, I cannot really say much except... How many VoIP guys were taking ak47 rounds while I was on top of the Iraqi Government building... A bit further in to the thread: Would you say that I am a productive member of the list and go pretty far out of my way to help people? Most of the time give useful info, like the Outbound Caller ID thread? [fish] And again: I do not email people... - then why did you just e-mail me off list? Your last post: ...because I own thousands of ounces of silver bullion... Everyone a winner [and not one relevant to the thread or the discussion]. Anyway - truth does indeed hurt mate. Grab yourself a Kleenex as I throw you back in to the pond. Goodbye. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Totaro Sent: 11 May 2011 17:36 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] When someone helps you, at least let them know if the problem is resolved or not I am not upset in least, well I am but that's because I own thousands of ounces of silver bullion and I am watching in get pummeled again. Good thing I bought the bulk of it when it was only $12 an ounce. http://www.kitco.com/charts/livesilver.html You are an angry person and it is sad. It is also sad that the example I requested earlier is something posted later. The only reason for that is because you had nothing to back up any of your rage. Seek help, please. If you feel like you want to hurt yourself or others, have yourself committed right away. I am serious. If you are voluntary, you can leave when you want. Thanks, Steve Totaro On Wed, May 11, 2011 at 12:13 PM, Andrew Thomas a...@datavox.co.uk wrote: Seems I have upset the God that is Steve Totaro! You want an example? OK - your last post. Has nothing to do with the thread (or our 'discussion') but yet you chose to post it as yet another self pat-on-the-back! I could produce a lot more - but you now bore me. You know it must be so hard being so perfect Steve. I so wish I was you! Have a really nice day :) From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Totaro Sent: 11 May 2011 16:38 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] When someone helps you, at least let them know if the problem is resolved or not Yeah, I am not sure why dude went on the offensive. Got emotional but could not produce a single example of the name calling and insults he was hurling at me. Here is an email I received a very short time ago. Sender and company's name have been removed. | to Steve show details 9:15 AM (2 hours ago) steve, I haven't been active in the * community for a while but ran across an interesting project that I would like to pursue. [COMPANY] in springfield needs a * admin part time and I could use the steady income plus I would like to get my hands back into *... I wanted to check with you first because this is your neck of the woods... do you have any experience with them? recently, I have been just lurking on the [asterisk-] lists. thanks for supporting the [asterisk-] groups in a big way. On Wed, May 11, 2011 at 11:28 AM, Sherwood McGowan sherwood.mcgo...@gmail.com wrote: Wow...somehow this turned into a something so much darker than the original intent*sits back and watches the show* Thanks guys, that little mini bonfire made an otherwise boring day into an entertaining Asterisk-Users version of WWE Raw. Cheers! Sherwood McGowan --
[asterisk-users] About minimum requirements to install PSTN GW+SIP Client
Dear all, Could you teach me minimum required Asterisk modules, application and etc to install PSTN GW+SIP Client functionalities in a PC as shown below? I have already downloaded astersik-1.8.3.3 and dahdi-linux-complete-2.4.1.2+2.4.1 in the PC. Analog telephone --- PSTN GW + SIP Client --- SIP Server PC Since available memory in the PC is very low, I have to select minimum requirements on screen Asterisk Module and Build Option Selection diplayed by executing make menuselect on console. First, I have selected chan_dahdi, chan_local and chan_sip as Channel Drivers on the screen, but don't understand that what are required in Applications, Resource Modules and etc on it to satisfy the above functionalities. Regards, Yagishita -- _ -- 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] Friday VUC: Discussion of Mobile SIP, Microsoft Lync
This should be interesting, a double header Friday at 12 Noon EDT, session 2 at 1PM EDT. 1) Pascal Doré, Media5corp. Pascal will talk about what they've been up to in the year since his last visit. Thanks to the Asterisk mailing list and VoIP community, their Media5fone was able to fix its g722 implementation. I like their product a lot and used it extensively on my old iPod Touch to make and receive phone calls on our server. SIP does rock when it works. 2) Dave Michels, VUC pillar member talks about Lync, good timing in light of Microsoft's purchase of Skype. Should be a lot of interesting commentary around the whole context. Join the call on sip:200...@login.zipdx.com or see http://vuc.me for all the call in options including: - GTalk voipusersconfere...@gmail.com - Skype bridge skype:vuc.me - live mp3 stream - IRC: #vuc on freenode.net - even PSTN and iNum VUC in your time zone: http://vuc.me/next Hope to hear you there! :r -- _ -- 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] When someone helps you, at least let them know if the problem is resolved or not
But it is all true. Don't rage on me if you are upset by my accomplishments. It was to establish the conditions I work under and the fact that I can and do solve almost all of my setbacks myself in very disparate setups, from VSAT, poorest countries in the world, war, explosions, getting shot at. I was there when the Ministry of Justice was bombed. I try everything under the sun and logically work out the the issue and get the job done. I have never expected someone to feed me the answers with a silver spoon. I am way too old school for that. To be honest, the list didn't help me very much except to the extent that I had to breakdown the problem and what was tried in words. Generally, after hitting send and re-reading my question, I answered it myself. It was the act of verbalizing and defining the problem rather than having it be something abstract in my mind. I don't try to tell people what to do about posting the solution It's all on the wheel, it all comes around. Don't hate. and don't let people live rent free in your head. Was all of this rage and hate necessary? Did I insult you in some way? I don't think I did any name calling or hating. Again, if you are depressed or feel like you want to hurt yourself or others, commit yourself, get help, the world will still be here when you are better able to deal with it. Don't lie to yourself and others, this was never meant to be this was intended as a bit of fun BS. http://www.drirene.com/thine_own.htm Thanks, Steve Totaro On Thu, May 12, 2011 at 3:01 AM, Andrew Thomas a...@datavox.co.uk wrote: [This is my last post in this thread - as I really CBA anymore!] Wow! You really don't see it do you? Fair enough. I thought you were just playing along with my 'ego baiting' game - but it seems I hit the mother load of all ego's here. Apologies to all 'watchers' - but this was intended as a bit of fun - and I thought Steve was just playing along. Seems I was wrong. As a parting gesture though - I'll give you an example from your first post in here (which reads more like a CV/resume than a post! [in fact, they all do]): I was the number 1 poster on this list a couple of years ago I don't really do job searches, I am usually offered a job or project and approached by the client. My last trip was to Iraq, but I have been to Senegal, Sierra Leone, Guinea, Ghana, Liberia to help rebuild the infrastructure for USAID. For the Dept of State, I set up... For DoD/Dos, I cannot really say much except... How many VoIP guys were taking ak47 rounds while I was on top of the Iraqi Government building... A bit further in to the thread: Would you say that I am a productive member of the list and go pretty far out of my way to help people? Most of the time give useful info, like the Outbound Caller ID thread? [fish] And again: I do not email people... - then why did you just e-mail me off list? Your last post: ...because I own thousands of ounces of silver bullion... Everyone a winner [and not one relevant to the thread or the discussion]. Anyway - truth does indeed hurt mate. Grab yourself a Kleenex as I throw you back in to the pond. Goodbye. -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Steve Totaro *Sent:* 11 May 2011 17:36 *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] When someone helps you, at least let them know if the problem is resolved or not I am not upset in least, well I am but that's because I own thousands of ounces of silver bullion and I am watching in get pummeled again. Good thing I bought the bulk of it when it was only $12 an ounce. http://www.kitco.com/charts/livesilver.html You are an angry person and it is sad. It is also sad that the example I requested earlier is something posted later. The only reason for that is because you had nothing to back up any of your rage. Seek help, please. If you feel like you want to hurt yourself or others, have yourself committed right away. I am serious. If you are voluntary, you can leave when you want. Thanks, Steve Totaro On Wed, May 11, 2011 at 12:13 PM, Andrew Thomas a...@datavox.co.ukwrote: Seems I have upset the God that is Steve Totaro! You want an example? OK - your last post. Has nothing to do with the thread (or our 'discussion') but yet you chose to post it as yet another self pat-on-the-back! I could produce a lot more - but you now bore me. You know it must be so hard being so perfect Steve. I so wish I was you! Have a really nice day :) -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Steve Totaro *Sent:* 11 May 2011 16:38 *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] When someone helps you, at least
Re: [asterisk-users] When someone helps you, at least let them know if the problem is resolved or not
This is beginning to turn even more unpleasant than the original breach of netiquette which prompted the discussion -- like one of those fights which starts with a raised voice, escalates to fisticuffs and then weapons, and by the time innocent bystanders are getting injured nobody can even remember the original point being argued. Please, let's all just move on -- and remember to (1) remove extraneous quoted material, because it saves bandwidth; (2) post *below* the points to which we are replying, because that is the natural flow of a conversation; and (3) acknowledge when issues are resolved with at least a quick Thank you, this worked for me -- both to show that no more need be said, and to indicate to future readers of the archives that a working solution was found. Call me an old romantic if you like, but I firmly believe that we owe it to future generations coming to read our posts in the mailing list archives, for them to find the archives as helpful as we found the posts we once had to read in the mailing list archives to get to the position of posting our own answers. -- AJS Answers come *after* questions. -- _ -- 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] When someone helps you, at least let them know if the problem is resolved or not
On Thu, May 12, 2011 at 4:28 AM, A J Stiles asterisk_l...@earthshod.co.ukwrote: This is beginning to turn even more unpleasant than the original breach of netiquette which prompted the discussion -- like one of those fights which starts with a raised voice, escalates to fisticuffs and then weapons, and by the time innocent bystanders are getting injured nobody can even remember the original point being argued. Please, let's all just move on -- and remember to (1) remove extraneous quoted material, because it saves bandwidth; (2) post *below* the points to which we are replying, because that is the natural flow of a conversation; and (3) acknowledge when issues are resolved with at least a quick Thank you, this worked for me -- both to show that no more need be said, and to indicate to future readers of the archives that a working solution was found. Call me an old romantic if you like, but I firmly believe that we owe it to future generations coming to read our posts in the mailing list archives, for them to find the archives as helpful as we found the posts we once had to read in the mailing list archives to get to the position of posting our own answers. -- AJS Answers come *after* questions. Agreed, you should be a Diplomat. I will always defend myself though. If they drop it, I will too. If I am attacked, I will defend. Thanks, Steve Totaro PS 42 is the answer, now what is the quesstion. :) -- _ -- 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] When someone helps you, at least let them know if the problem is resolved or not
PS 42 is the answer, now what is the quesstion. :) What is the difference between a bird? -- _ -- 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] multiple calls into hold
Hi list, Is there any way by which we can put multiple calls into hold with asterisk. like A to B. then C to B and A on hold. then D to B now C ,A on hold like wise.. -- - Thanks and regards Virendra Bhati +91-9172341457 Asterisk Engineer -- _ -- 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] multiple calls into hold
On Thu, 12 May 2011, virendra bhati wrote: Hi list, Is there any way by which we can put multiple calls into hold with asterisk. like A to B. then C to B and A on hold. then D to B now C ,A on hold like wise.. It's more a phone issue that asterisk. Just get a multi-line phone (GXP2000, Snom 360, etc.) and enable call-waiting on it. Then you can shuttle between calls, transfers, etc. Just like a real receptionist ;) Gordon -- _ -- 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] Higher CPU usage on 1.6.1 than 1.4?
Hi David, When I was testing 1.6.1 for high volume channels, I couldn't get over 1000 channels / 40 CPS without the load average spiking up due to io wait. I switched back to 1.4 and I can go to 3000 channels / 75 CPS with no io wait and a load average in the 1s. It seemed like it was caused by the new timing system in 1.6.1 even though I wasn't proxying media using only SIP. I haven't tried 1.8 yet to see if it handles large call volumes any better. ~Jared On Wed, May 11, 2011 at 8:29 PM, David Cunningham dcunning...@voisonics.com wrote: Hello, We have a customer who upgraded from Asterisk 1.4 to 1.6.1.22 and is now experiencing higher CPU utilization on their server. I can't see anything wrong, so is this just expected with 1.6? Can anyone help explain it? Thanks for any advice. -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- 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] multiple calls into hold
Hi , I am using Cisco 7940/60 phone. Is this okay or we need another phone for that. plz suggest me ' On Thu, May 12, 2011 at 3:52 PM, Gordon Henderson gordon+aster...@drogon.net wrote: On Thu, 12 May 2011, virendra bhati wrote: Hi list, Is there any way by which we can put multiple calls into hold with asterisk. like A to B. then C to B and A on hold. then D to B now C ,A on hold like wise.. It's more a phone issue that asterisk. Just get a multi-line phone (GXP2000, Snom 360, etc.) and enable call-waiting on it. Then you can shuttle between calls, transfers, etc. Just like a real receptionist ;) Gordon -- _ -- 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 -- - Thanks and regards Virendra Bhati +91-9172341457 Asterisk Engineer -- _ -- 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] Trying out a new version with sangoma card
In asterisk CLI do pri show spans. The fact the card is in RED alert means the hardware does not see the pri line connected to the card. I probably made a mistake in copying / pasting. pri show spans was showing something like : PRI span 1/0: Provisioned, Up, Active Calls can enter, I see them arriving on the console, but they imediatly got hangun, cause 6. -- _ -- 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] multiple calls into hold
On Thu, 12 May 2011, virendra bhati wrote: On Thu, May 12, 2011 at 3:52 PM, Gordon Henderson gordon+aster...@drogon.net wrote: On Thu, 12 May 2011, virendra bhati wrote: Hi list, Is there any way by which we can put multiple calls into hold with asterisk. like A to B. then C to B and A on hold. then D to B now C ,A on hold like wise.. It's more a phone issue that asterisk. Just get a multi-line phone (GXP2000, Snom 360, etc.) and enable call-waiting on it. Then you can shuttle between calls, transfers, etc. Just like a real receptionist ;) Hi , I am using Cisco 7940/60 phone. Is this okay or we need another phone for that. plz suggest me ' I have no idea. You could always google it and read the manual. However I've done this for you (I'll send the bill shortly). It looks like it's designed as a generic office user type phone rather than for reception type use. http://www.cisco.com/en/US/products/hw/phones/ps379/ps1854/index.html it says: The Cisco Unified IP Phone 7940G is well suited for employees in a basic office cubicle environment--such as transaction type workers--who conduct a moderate amount of business by phone. Gordon -- _ -- 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] concurrent call tracking
Hi, this is my first post to mailing list, so sorry in case i'm doing something wrong. when i want to count concurent calls from particular user, i dont use any cron jobs or counters in dialplan, run query on cdr, something like: SEELCT dst, calldate, IF(action = 'substract', @count := @count - 1, @count := @count + 1) FROM (SELECT dst, calldate, 'substract' AS 'action' FROM cdr WHERE calldate between '2011.05.12' AND '2011.05.13' AND src = 500 UNION SELECT dst, DATE_ADD(calldate, INTERVAL duration SECOND), 'add' FROM cdr WHERE calldate between '2011.05.12' AND '2011.05.13' AND src = 500) JOIN (SELECT @count := 0) ORDER BY calldate; - Original Message - From: Skyler To: asterisk-users@lists.digium.com Sent: Wednesday, May 11, 2011 19:57 Subject: [asterisk-users] concurrent call tracking Hi all, I would like to track/store concurrent call usage per user by day/week/month and get server totals by day/week/month. Google comes up with mostly info regarding concurrent call limits, though my goal is to calculate actual concurrent channel usage and add it into reporting. I'm using * 1.6.2 + mysql - realtime (no gui). Any suggestions / open-source / AGI on where to start looking into implementing something like this? TIA, Skyler -- _ -- 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] Trying out a new version with sangoma card
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nicolas Ross Sent: Thursday, May 12, 2011 6:36 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Trying out a new version with sangoma card In asterisk CLI do pri show spans. The fact the card is in RED alert means the hardware does not see the pri line connected to the card. I probably made a mistake in copying / pasting. pri show spans was showing something like : PRI span 1/0: Provisioned, Up, Active Calls can enter, I see them arriving on the console, but they imediatly got hangun, cause 6. Show us the output of a failed call with pri debug enabled on that span. -- _ -- 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] Higher CPU usage on 1.6.1 than 1.4?
Jared, Thank you for that information! Has anyone else had an experience like this? On 12 May 2011 20:25, Jared Geiger compuw...@gmail.com wrote: Hi David, When I was testing 1.6.1 for high volume channels, I couldn't get over 1000 channels / 40 CPS without the load average spiking up due to io wait. I switched back to 1.4 and I can go to 3000 channels / 75 CPS with no io wait and a load average in the 1s. It seemed like it was caused by the new timing system in 1.6.1 even though I wasn't proxying media using only SIP. I haven't tried 1.8 yet to see if it handles large call volumes any better. ~Jared On Wed, May 11, 2011 at 8:29 PM, David Cunningham dcunning...@voisonics.com wrote: Hello, We have a customer who upgraded from Asterisk 1.4 to 1.6.1.22 and is now experiencing higher CPU utilization on their server. I can't see anything wrong, so is this just expected with 1.6? Can anyone help explain it? Thanks for any advice. -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- 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 -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -- _ -- 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] Trying out a new version with sangoma card
Show us the output of a failed call with pri debug enabled on that span. It will be difficult, since the PRI is in use on our old asterisk box. I will have to get to the colo at night, to avoid disrupting calls during the day. Is there any other thing that I should collect ? -- _ -- 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] Different IP addresss for SIP and RTP
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of mayamatakeshi Sent: Thursday, May 12, 2011 12:58 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Different IP addresss for SIP and RTP Hello, is it possible to set an IP address for RTP different than the one used for SIP? I want to use asterisk behind a sip proxy (opensips), but I was thinking if I could avoid having to run rtpproxy on the sip proxy server and let asterisk itself take care of it. So that: Asterisk SIP address : local ip address Asterisk RTP address : global ip address regards, takeshi [Danny Nicholas] You handicap potential responder by not stating your Asterisk release - that being said, try putting bindaddr=global.ip.addr in rtp.conf and see if that works for you. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] log full of Name or service not known
Hi! Here's a user with mobile phone - however why does it treat this as ERROR ? I have a log full of that --- -- Registered SIP '0010106' at 212.93.100.181:3698 [2011-05-12 16:07:57] NOTICE[30258]: chan_sip.c:19679 handle_response_peerpoke: Peer '0010106' is now Reachable. (212ms / 1ms) [2011-05-12 16:07:57] ERROR[30258]: netsock2.c:245 ast_sockaddr_resolve: getaddrinfo(212.93.100.181:3698, 3698, ...): Name or service not known -- _ -- 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] concurrent call tracking
On 11-05-11 06:36 PM, Skyler wrote: Thanks Dovid, if you don't mind sharing the code and the dial plan side I'd like to take a look at it for sure. The dial plan example Leif replied with is pretty much what I was thinking, just didn't have a clue how to go about it. ;) You could also look into using LOCK() and UNLOCK() dialplan applications to make sure each insert happens sequentially. Leif. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8.4 Now Available
On 11-05-11 09:31 PM, Jose P. Espinal wrote: Download links on the website have not been updated (asterisk.org) Oops sorry! I will fix that right.. now! Leif. -- _ -- 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] ConfBridge for 1.8 ?
Hey Guys! I am looking ConfBridge for 1.8 version of asterisk. How could i obtain and install with 1.8 ? -S -- _ -- 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] ConfBridge for 1.8 ?
Holly Cow! Its there already sorry i thought it will only comes with 1.10. We are using meetme since last 5 year do you think confbridge is better then meetme ? just need your suggestion /usr/lib/asterisk/modules/app_confbridge.so From: satish...@hotmail.com To: asterisk-users@lists.digium.com Date: Thu, 12 May 2011 14:33:12 + Subject: [asterisk-users] ConfBridge for 1.8 ? Hey Guys! I am looking ConfBridge for 1.8 version of asterisk. How could i obtain and install with 1.8 ? -S -- _ -- 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] ConfBridge for 1.8 ?
On 05/12/2011 09:37 AM, satish patel wrote: Holly Cow! Its there already sorry i thought it will only comes with 1.10. We are using meetme since last 5 year do you think confbridge is better then meetme ? just need your suggestion /usr/lib/asterisk/modules/app_confbridge.so The app_confbridge in Asterisk 1.8 is very different from the one in trunk (what will become Asterisk 1.10). -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ConfBridge for 1.8 ?
Thanks Kevin, Good to know. Different mean features vise or performance ? Do you think it is a good idea to replace meetme with confbridge in current 1.8 or i should wait for 1.10 ? -S Date: Thu, 12 May 2011 09:50:12 -0500 From: kpflem...@digium.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] ConfBridge for 1.8 ? On 05/12/2011 09:37 AM, satish patel wrote: Holly Cow! Its there already sorry i thought it will only comes with 1.10. We are using meetme since last 5 year do you think confbridge is better then meetme ? just need your suggestion /usr/lib/asterisk/modules/app_confbridge.so The app_confbridge in Asterisk 1.8 is very different from the one in trunk (what will become Asterisk 1.10). -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] test call generator
Hello Everyone, I wonder if someone could share a manual about using SIPp for Asterisk's testing. I'll be gratefull Regards, Elder Arohuanca Lima - Peru On Tue, Sep 30, 2008 at 12:59 PM, zac wolfe zac.wo...@gmail.com wrote: Sipp looks pretty good! I don't know how I missed this one. This would've saved me tons of time a couple months ago. I plan on using it to load test using 2 Asterisk servers, one to initiate the SIP calls, the other to receive. Thanks for the tip Alex. Zac Wolfe Safi Systems LLC www.safisystems.com On Sat, Sep 27, 2008 at 5:58 AM, Alex Balashov abalas...@evaristesys.comwrote: What you are looking for is SIPP: http://sipp.sourceforge.net/ It won't intrinsically tell you anything about the data; it's up to you to appropriate the findings. But it accomplishes the generation of traffic (and dummy media!) on a technical level. Igor Hernandez wrote: Sam Tam wrote: Hello everyone I am trying to look for a free test call generator that will get me some stats like PDD, ASR and call quality etc on each route. As well as do test at every interval too If you know something like this please enlighten me. Sam ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hey Sam, I've been looking for such a tool also. I can't seem to find a tool that does those things. If nothing comes up in the next couple of weeks I'm going to code something up, I wouldn't mind letting you and anyone else who might be interested have the source once its done. Let me know if you find anything thats already out there in the meantime, might just save me a few hours of work. Regards, -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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] Light indicator managed by Asterisk
Hello, is there some way to make Asterisk light up a certain light on an IP-phone ? Like MWI, the message waiting indicator can light up if there is voicemail. Could this light, or even other lights (like BLF-buttons) be used to give a visual notification to the user ? For example : if a certain value is set in the Mysql-DB and Asterisk reads out this value, can Asterisk react upon it inside the dialplan to make a light lit up ? 2nd example : if a certain extension is called, can we perform inside the dialplan an action that makes a light lit up on a Snom or Yealink IP-phone ? I don't know if all this is at all possible, but it doesn't harm asking I guess... If BLF works, then maybe more things are possible in the same way. Just thinking outside the box here. Kind regards, Jonas. -- _ -- 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] Realtime - ara180
Hi all,
A week or so down the list, i read that not many people were using
realtime on an Asterisk18, so i had this afternoon a go at it...
[sorry for the inconveneant line-wraps]
First i did:
mysql create database asterisk;
mysql grant all on asterisk.* to 'voipadmin'@'localhost' identified by
next i used the info from the wiki:
CREATE TABLE `sip_devices` (
`id` int(11) NOT NULL AUTO_INCREMENT,
`name` varchar(80) NOT NULL DEFAULT '', `context` varchar(80) DEFAULT
NULL,
[and lots more]
and populated it with a test user:
mysql 'secret'; insert into sip_devices (name, secret, username, host,
nat) values ('0031756', 'geheim', '0031756', 'dynamic', 'Yes');
tested the database:
mysql -h localhost -u voipadmin -p
Enter password:
Welcome to the MySQL monitor. Commands end with ; or \g.
Your MySQL connection id is 12645
Server version: 5.0.67 SUSE MySQL RPM
Type 'help;' or '\h' for help. Type '\c' to clear the buffer.
mysql select * from sip_devices\G
ERROR 1046 (3D000): No database selected
mysql use asterisk;
Database changed
mysql select name,username,secret,host,nat from sip_devices;
+-+-++-+-+
| name| username| secret | host| nat |
+-+-++-+-+
| 0031756 | 0031756 | geheim | dynamic | Yes |
+-+-++-+-+
1 row in set (0.00 sec)
mysql
So-far all looked, as expected, great
Changed the *-config files:
(method, db-name, table-name)
;
;iaxusers = odbc,asterisk
;iaxpeers = odbc,asterisk
;sipusers = odbc,asterisk
sipusers = mysql,asterisk,sip_devices
sippeers = mysql,asterisk,sip_devices
;sippeers = odbc,asterisk
;sipregs = odbc,asterisk
;voicemail = odbc,asterisk
And restarted the asterisk-process...
Some lines from /var/log/asterisk/messages:
May 12 14:05:33] WARNING[2585] config.c: Realtime mapping for 'sippeers'
found to engine 'mysql', but the engine is not available
[May 12 14:05:33] WARNING[2585] config.c: Realtime mapping for
'sippeers' found to engine 'mysql', but the engine is not available
[May 12 14:05:55] WARNING[2630] config.c: Realtime mapping for
'sippeers' found to engine 'mysql', but the engine is not available
And on the asterisk console:
kc3054*CLI sip show peers
Name/username Host Dyn Forcerport ACL Port Status
0277611 (Unspecified) D N 0Unmonitored
j.witvliet (Unspecified) D N 0Unmonitored
2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 2
offline] [May 12 14:17:47] WARNING[2630]: config.c:2045 find_engine:
Realtime mapping for 'sippeers' found to engine 'mysql', but the engine
is not available
CLI sip show users
Username Secret Accountcode Def.Context ACL ForcerPort
j.witvliet geheim default No Yes
0277611 25b06d3a0b5ef73 default No Yes
CLI
1) as shown above, access to mysql seems to be OK,
2) * did not complain at sip-show-users but only for sip-show-peers
fyi, i use: asterisk180-1.8.3.2-87.1.x86_64.rpm
Any suggesions?
Hans
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Re: [asterisk-users] Light indicator managed by Asterisk
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: Thursday, May 12, 2011 11:50 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Light indicator managed by Asterisk Hello, is there some way to make Asterisk light up a certain light on an IP-phone ? Like MWI, the message waiting indicator can light up if there is voicemail. Could this light, or even other lights (like BLF-buttons) be used to give a visual notification to the user ? For example : if a certain value is set in the Mysql-DB and Asterisk reads out this value, can Asterisk react upon it inside the dialplan to make a light lit up ? 2nd example : if a certain extension is called, can we perform inside the dialplan an action that makes a light lit up on a Snom or Yealink IP-phone ? I don't know if all this is at all possible, but it doesn't harm asking I guess... If BLF works, then maybe more things are possible in the same way. Just thinking outside the box here. Kind regards, Jonas. [Danny Nicholas] The answer is highly dependent on flavor of Asterisk and phone. In my experience, Asterisk can generally only light up the MWI light on any applicable phone. I have Polycom 501 phones and I can make MWI light up by leaving a message for that phone or my generic extension 108. If I zap the message, the light goes off. So in either of your examples, I could call an AGI to turn the light on or off. As I read, SNOM are pretty much equivalent to Polycoms regarding MWI functionality, don't know about Yealink. -- _ -- 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] Light indicator managed by Asterisk
On Thu, May 12, 2011 at 12:50 PM, Jonas Kellens jonas.kell...@telenet.be wrote: Hello, is there some way to make Asterisk light up a certain light on an IP-phone ? Like MWI, the message waiting indicator can light up if there is voicemail. Could this light, or even other lights (like BLF-buttons) be used to give a visual notification to the user ? For example : if a certain value is set in the Mysql-DB and Asterisk reads out this value, can Asterisk react upon it inside the dialplan to make a light lit up ? 2nd example : if a certain extension is called, can we perform inside the dialplan an action that makes a light lit up on a Snom or Yealink IP-phone ? I don't know if all this is at all possible, but it doesn't harm asking I guess... If BLF works, then maybe more things are possible in the same way. Just thinking outside the box here. Kind regards, Jonas. On snom and other phones it is easy... http://wiki.snom.com/Interoperability/PBX/Asterisk#Extension_Monitoring_.28BLF.29_.26_Call_Pick-Up Also look at SLA http://svn.asterisk.org/svn/asterisk/trunk/configs/sla.conf.sample -- ~~~ Andrew lathama Latham lath...@gmail.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] Light indicator managed by Asterisk
On Thu, 2011-05-12 at 18:50 +0200, Jonas Kellens wrote: Hello, is there some way to make Asterisk light up a certain light on an IP-phone ? Like MWI, the message waiting indicator can light up if there is voicemail. Could this light, or even other lights (like BLF-buttons) be used to give a visual notification to the user ? For example : if a certain value is set in the Mysql-DB and Asterisk reads out this value, can Asterisk react upon it inside the dialplan to make a light lit up ? 2nd example : if a certain extension is called, can we perform inside the dialplan an action that makes a light lit up on a Snom or Yealink IP-phone ? I don't know if all this is at all possible, but it doesn't harm asking I guess... If BLF works, then maybe more things are possible in the same way. Just thinking outside the box here. BLF lights can be manipulated with Hints and the DEVSTATE function to set custom device states. This way you can have a BLF light react to any event you want. The MWI can be manipulated in several ways. Last week someone asked this question and got several answers. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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] Different IP addresss for SIP and RTP
On Thu, May 12, 2011 at 10:07 PM, Danny Nicholas da...@debsinc.com wrote: -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *mayamatakeshi *Sent:* Thursday, May 12, 2011 12:58 AM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] Different IP addresss for SIP and RTP Hello, is it possible to set an IP address for RTP different than the one used for SIP? I want to use asterisk behind a sip proxy (opensips), but I was thinking if I could avoid having to run rtpproxy on the sip proxy server and let asterisk itself take care of it. So that: Asterisk SIP address : local ip address Asterisk RTP address : global ip address regards, takeshi *[Danny Nicholas] * *You handicap potential responder by not stating your Asterisk release – that being said, try putting bindaddr=global.ip.addr in rtp.conf and see if that works for you.* Hello, I have installed Asterisk 1.8. I've tried using parameter bindaddr but it didn't work. Looking at the code that reads rtp.conf I could not locate any place indicating that this address could be set: http://svn.asterisk.org/svn/asterisk/branches/1.8/res/res_rtp_asterisk.c http://svn.asterisk.org/svn/asterisk/trunk/res/res_rtp_asterisk.c So I think this is not currently possible. Thanks anyway. regards, takeshi -- _ -- 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] Light indicator managed by Asterisk
On 05/12/2011 06:58 PM, Andrew Latham wrote: On Thu, May 12, 2011 at 12:50 PM, Jonas Kellens jonas.kell...@telenet.be wrote: Hello, is there some way to make Asterisk light up a certain light on an IP-phone ? Like MWI, the message waiting indicator can light up if there is voicemail. Could this light, or even other lights (like BLF-buttons) be used to give a visual notification to the user ? For example : if a certain value is set in the Mysql-DB and Asterisk reads out this value, can Asterisk react upon it inside the dialplan to make a light lit up ? 2nd example : if a certain extension is called, can we perform inside the dialplan an action that makes a light lit up on a Snom or Yealink IP-phone ? I don't know if all this is at all possible, but it doesn't harm asking I guess... If BLF works, then maybe more things are possible in the same way. Just thinking outside the box here. Kind regards, Jonas. On snom and other phones it is easy... http://wiki.snom.com/Interoperability/PBX/Asterisk#Extension_Monitoring_.28BLF.29_.26_Call_Pick-Up Also look at SLA http://svn.asterisk.org/svn/asterisk/trunk/configs/sla.conf.sample I know how MWI and BLF works, that's not my question. I'm talking about lighting up lights on the phone on other occasions, not related to MWI or BLF. Check the examples I gave again. Can Asterisk (through SIP or maybe another protocol) control lights on an IP-phone like Snom, Yealink or other ? Also : can we control SIP notify or SIP option packets inside the dialplan ? Kind regards, Jonas. -- _ -- 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] concurrent call tracking
Many thanks to all that replied. I'm going to test out the suggestions/scenarios and I'll post back with what worked for me. S. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leif Madsen Sent: Thursday, May 12, 2011 6:29 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] concurrent call tracking On 11-05-11 06:36 PM, Skyler wrote: Thanks Dovid, if you don't mind sharing the code and the dial plan side I'd like to take a look at it for sure. The dial plan example Leif replied with is pretty much what I was thinking, just didn't have a clue how to go about it. ;) You could also look into using LOCK() and UNLOCK() dialplan applications to make sure each insert happens sequentially. Leif. -- _ -- 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 _ No virus found in this message. Checked by AVG - www.avg.com Version: 10.0.1325 / Virus Database: 1500/3632 - Release Date: 05/11/11 -- _ -- 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] Light indicator managed by Asterisk
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: Thursday, May 12, 2011 12:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Light indicator managed by Asterisk On 05/12/2011 06:58 PM, Andrew Latham wrote: On Thu, May 12, 2011 at 12:50 PM, Jonas Kellens jonas.kell...@telenet.be wrote: Hello, is there some way to make Asterisk light up a certain light on an IP- phone ? Like MWI, the message waiting indicator can light up if there is voicemail. Could this light, or even other lights (like BLF-buttons) be used to give a visual notification to the user ? For example : if a certain value is set in the Mysql-DB and Asterisk reads out this value, can Asterisk react upon it inside the dialplan to make a light lit up ? 2nd example : if a certain extension is called, can we perform inside the dialplan an action that makes a light lit up on a Snom or Yealink IP- phone ? I don't know if all this is at all possible, but it doesn't harm asking I guess... If BLF works, then maybe more things are possible in the same way. Just thinking outside the box here. Kind regards, Jonas. On snom and other phones it is easy... http://wiki.snom.com/Interoperability/PBX/Asterisk#Extension_Monitoring_.2 8BLF.29_.26_Call_Pick-Up Also look at SLA http://svn.asterisk.org/svn/asterisk/trunk/configs/sla.conf.sample I know how MWI and BLF works, that's not my question. I'm talking about lighting up lights on the phone on other occasions, not related to MWI or BLF. Check the examples I gave again. Can Asterisk (through SIP or maybe another protocol) control lights on an IP-phone like Snom, Yealink or other ? Also : can we control SIP notify or SIP option packets inside the dialplan ? Kind regards, Jonas. [Danny Nicholas] stabbing #1 I doubt it, read your phone documentation #2 using special contexts and addSipHeader commands, perhaps?/stabbing -- _ -- 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] Light indicator managed by Asterisk
On 05/12/2011 07:12 PM, Carlos Chavez wrote: On Thu, 2011-05-12 at 18:50 +0200, Jonas Kellens wrote: Hello, is there some way to make Asterisk light up a certain light on an IP-phone ? Like MWI, the message waiting indicator can light up if there is voicemail. Could this light, or even other lights (like BLF-buttons) be used to give a visual notification to the user ? For example : if a certain value is set in the Mysql-DB and Asterisk reads out this value, can Asterisk react upon it inside the dialplan to make a light lit up ? 2nd example : if a certain extension is called, can we perform inside the dialplan an action that makes a light lit up on a Snom or Yealink IP-phone ? I don't know if all this is at all possible, but it doesn't harm asking I guess... If BLF works, then maybe more things are possible in the same way. Just thinking outside the box here. BLF lights can be manipulated with Hints and the DEVSTATE function to set custom device states. This way you can have a BLF light react to any event you want. This means that extensions/hints need to be defined to be able to control a BLF-light that monitors this extension ? I agree that this gives some control over a light/button on an IP-phone. The MWI can be manipulated in several ways. Last week someone asked this question and got several answers. You don't perhaps have a link to the discussion ? I don't really follow this list constantly so I've certainly missed out on this subject. Kind regards, Jonas. -- _ -- 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] Light indicator managed by Asterisk
On 05/12/2011 07:24 PM, Danny Nicholas wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: Thursday, May 12, 2011 12:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Light indicator managed by Asterisk On 05/12/2011 06:58 PM, Andrew Latham wrote: On Thu, May 12, 2011 at 12:50 PM, Jonas Kellens jonas.kell...@telenet.be wrote: Hello, is there some way to make Asterisk light up a certain light on an IP- phone ? Like MWI, the message waiting indicator can light up if there is voicemail. Could this light, or even other lights (like BLF-buttons) be used to give a visual notification to the user ? For example : if a certain value is set in the Mysql-DB and Asterisk reads out this value, can Asterisk react upon it inside the dialplan to make a light lit up ? 2nd example : if a certain extension is called, can we perform inside the dialplan an action that makes a light lit up on a Snom or Yealink IP- phone ? I don't know if all this is at all possible, but it doesn't harm asking I guess... If BLF works, then maybe more things are possible in the same way. Just thinking outside the box here. Kind regards, Jonas. On snom and other phones it is easy... http://wiki.snom.com/Interoperability/PBX/Asterisk#Extension_Monitoring_.2 8BLF.29_.26_Call_Pick-Up Also look at SLA http://svn.asterisk.org/svn/asterisk/trunk/configs/sla.conf.sample I know how MWI and BLF works, that's not my question. I'm talking about lighting up lights on the phone on other occasions, not related to MWI or BLF. Check the examples I gave again. Can Asterisk (through SIP or maybe another protocol) control lights on an IP-phone like Snom, Yealink or other ? Also : can we control SIP notify or SIP option packets inside the dialplan ? Kind regards, Jonas. [Danny Nicholas] stabbing #1 I doubt it, read your phone documentation #2 using special contexts and addSipHeader commands, perhaps?/stabbing So if the IP-phone lets me program to react a certain way upon the presence of a custom SIP-header, then this could be implemented ? Humm.. maybe you're right. You don't perhaps know which phone type/brand gives me that much freedom ? Kind regards, Jonas. -- _ -- 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] Light indicator managed by Asterisk
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: Thursday, May 12, 2011 12:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Light indicator managed by Asterisk On 05/12/2011 07:12 PM, Carlos Chavez wrote: On Thu, 2011-05-12 at 18:50 +0200, Jonas Kellens wrote: Hello, is there some way to make Asterisk light up a certain light on an IP-phone ? Like MWI, the message waiting indicator can light up if there is voicemail. Could this light, or even other lights (like BLF-buttons) be used to give a visual notification to the user ? For example : if a certain value is set in the Mysql-DB and Asterisk reads out this value, can Asterisk react upon it inside the dialplan to make a light lit up ? 2nd example : if a certain extension is called, can we perform inside the dialplan an action that makes a light lit up on a Snom or Yealink IP-phone ? I don't know if all this is at all possible, but it doesn't harm asking I guess... If BLF works, then maybe more things are possible in the same way. Just thinking outside the box here. BLF lights can be manipulated with Hints and the DEVSTATE function to set custom device states. This way you can have a BLF light react to any event you want. This means that extensions/hints need to be defined to be able to control a BLF-light that monitors this extension ? I agree that this gives some control over a light/button on an IP-phone. The MWI can be manipulated in several ways. Last week someone asked this question and got several answers. You don't perhaps have a link to the discussion ? I don't really follow this list constantly so I've certainly missed out on this subject. Kind regards, Jonas. [Danny Nicholas] Try this http://lists.digium.com/pipermail/asterisk-users/2011-May/262062.html -- _ -- 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] Light indicator managed by Asterisk
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: Thursday, May 12, 2011 12:31 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Light indicator managed by Asterisk On 05/12/2011 07:24 PM, Danny Nicholas wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: Thursday, May 12, 2011 12:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Light indicator managed by Asterisk On 05/12/2011 06:58 PM, Andrew Latham wrote: On Thu, May 12, 2011 at 12:50 PM, Jonas Kellens jonas.kell...@telenet.be wrote: Hello, is there some way to make Asterisk light up a certain light on an IP- phone ? Like MWI, the message waiting indicator can light up if there is voicemail. Could this light, or even other lights (like BLF-buttons) be used to give a visual notification to the user ? For example : if a certain value is set in the Mysql-DB and Asterisk reads out this value, can Asterisk react upon it inside the dialplan to make a light lit up ? 2nd example : if a certain extension is called, can we perform inside the dialplan an action that makes a light lit up on a Snom or Yealink IP- phone ? I don't know if all this is at all possible, but it doesn't harm asking I guess... If BLF works, then maybe more things are possible in the same way. Just thinking outside the box here. Kind regards, Jonas. On snom and other phones it is easy... http://wiki.snom.com/Interoperability/PBX/Asterisk#Extension_Monitoring_.2 8BLF.29_.26_Call_Pick-Up Also look at SLA http://svn.asterisk.org/svn/asterisk/trunk/configs/sla.conf.sample I know how MWI and BLF works, that's not my question. I'm talking about lighting up lights on the phone on other occasions, not related to MWI or BLF. Check the examples I gave again. Can Asterisk (through SIP or maybe another protocol) control lights on an IP-phone like Snom, Yealink or other ? Also : can we control SIP notify or SIP option packets inside the dialplan ? Kind regards, Jonas. [Danny Nicholas] stabbing #1 I doubt it, read your phone documentation #2 using special contexts and addSipHeader commands, perhaps?/stabbing So if the IP-phone lets me program to react a certain way upon the presence of a custom SIP-header, then this could be implemented ? Humm.. maybe you're right. You don't perhaps know which phone type/brand gives me that much freedom ? Kind regards, Jonas. [Danny Nicholas] Sorry I only know about Polycom and not as much about that as I'd like. -- _ -- 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] Light indicator managed by Asterisk
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Jonas Kellens
Sent: Thursday, May 12, 2011 1:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Light indicator managed by Asterisk
On 05/12/2011 06:58 PM, Andrew Latham wrote:
On Thu, May 12, 2011 at 12:50 PM, Jonas Kellens
jonas.kell...@telenet.be wrote:
Hello,
is there some way to make Asterisk light up a certain
light on an IP-phone ?
Like MWI, the message waiting indicator can light up if
there is voicemail.
Could this light, or even other lights (like BLF-buttons)
be used to give a
visual notification to the user ?
For example : if a certain value is set in the Mysql-DB
and Asterisk reads
out this value, can Asterisk react upon it inside the
dialplan to make a
light lit up ?
2nd example : if a certain extension is called, can we
perform inside the
dialplan an action that makes a light lit up on a Snom or
Yealink IP-phone ?
I don't know if all this is at all possible, but it
doesn't harm asking I
guess...
If BLF works, then maybe more things are possible in the
same way. Just
thinking outside the box here.
Kind regards,
Jonas.
On snom and other phones it is easy...
http://wiki.snom.com/Interoperability/PBX/Asterisk#Extension_M
onitoring_.28BLF.29_.26_Call_Pick-Up
Also look at SLA
http://svn.asterisk.org/svn/asterisk/trunk/configs/sla.conf.sample
I know how MWI and BLF works, that's not my question. I'm
talking about
lighting up lights on the phone on other occasions, not
related to MWI
or BLF. Check the examples I gave again.
Can Asterisk (through SIP or maybe another protocol)
control lights on
an IP-phone like Snom, Yealink or other ?
Also : can we control SIP notify or SIP option packets inside
the dialplan ?
This is all that I am aware of to control the phone lights.
pbx*CLI core show function DEVICE_STATE
-= Info about function 'DEVICE_STATE' =-
[Synopsis]
Get or Set a device state.
[Description]
The DEVICE_STATE function can be used to retrieve the device state from any
device state provider. For example:
NoOp(SIP/mypeer has state ${DEVICE_STATE(SIP/mypeer)})
NoOp(Conference number 1234 has state ${DEVICE_STATE(MeetMe:1234)})
The DEVICE_STATE function can also be used to set custom device state from
the dialplan. The 'Custom:' prefix must be used. For example:
Set(DEVICE_STATE(Custom:lamp1)=BUSY)
Set(DEVICE_STATE(Custom:lamp2)=NOT_INUSE)
You can subscribe to the status of a custom device state using a hint in
the dialplan:
exten = 1234,hint,Custom:lamp1
The possible values for both uses of this function are:
UNKNOWN | NOT_INUSE | INUSE | BUSY | INVALID | UNAVAILABLE | RINGING |
RINGINUSE | ONHOLD
[Syntax]
DEVICE_STATE(device)
[Arguments]
Not available
[See Also]
Not available
--
_
-- 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] Light indicator managed by Asterisk
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: Thursday, May 12, 2011 1:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Light indicator managed by Asterisk On 05/12/2011 07:12 PM, Carlos Chavez wrote: On Thu, 2011-05-12 at 18:50 +0200, Jonas Kellens wrote: Hello, is there some way to make Asterisk light up a certain light on an IP-phone ? Like MWI, the message waiting indicator can light up if there is voicemail. Could this light, or even other lights (like BLF-buttons) be used to give a visual notification to the user ? For example : if a certain value is set in the Mysql-DB and Asterisk reads out this value, can Asterisk react upon it inside the dialplan to make a light lit up ? 2nd example : if a certain extension is called, can we perform inside the dialplan an action that makes a light lit up on a Snom or Yealink IP-phone ? I don't know if all this is at all possible, but it doesn't harm asking I guess... If BLF works, then maybe more things are possible in the same way. Just thinking outside the box here. BLF lights can be manipulated with Hints and the DEVSTATE function to set custom device states. This way you can have a BLF light react to any event you want. This means that extensions/hints need to be defined to be able to control a BLF-light that monitors this extension ? I agree that this gives some control over a light/button on an IP-phone. The MWI can be manipulated in several ways. Last week someone asked this question and got several answers. You don't perhaps have a link to the discussion ? I don't really follow this list constantly so I've certainly missed out on this subject. pbx*CLI core show application minivmmwi -= Info about application 'MinivmMWI' =- [Synopsis] Send Message Waiting Notification to subscriber(s) of mailbox. [Description] This application is part of the Mini-Voicemail system, configured in min ivm.conf. MinivmMWI is used to send message waiting indication to any devices whose channels have subscribed to the mailbox passed in the first parameter. [Syntax] MinivmMWI(username@domain,urgent,new,old) [Arguments] username Voicemail username domain Voicemail domain urgent Number of urgent messages in mailbox. new Number of new messages in mailbox. old Number of old messages in mailbox. [See Also] Not available -- _ -- 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] Problem with PSTN calls (Asterisk as SIP client on embedded device)
Hi I've spent two days trying to solve this issue but to no prevail and I'm hoping to get some help. I've configured Asterisk as a SIP client, running on OpenWRT on an embedded device with onboard FXS and ATA. Asterisk is connecting to an external SIP provider on the Internet who in turn provides a PSTN gateway. I'm able to make calls to other SIP accounts registered on the same server who are outside my LAN. However, I can not make calls to any PSTN numbers. When trying to make PSTN calls it sounds like the person at the other end is immediately rejecting the call although I know this is not the case. Firstly, I'm absolutely sure that the PSTN gateway is working because I can make outbound PSTN calls with the same SIP account using other SIP clients (Empathy-SIP, SIPDroid) from the same LAN. However, when registering the same SIP account using Asterisk from OpenWRT all PSTN calls fail. Inbound calls from PSTN numbers also fail while calls from other SIP clients on the same server work fine. Thus, I'm fairly confident the problem is with my Asterisk configuration. The SIP accounts shows as registered in Asterisk. I've attached detailed error logs. The log files 'messages-pstn.log' shows the failed (PSTN) call and 'messages-voip.log' shows the successful (VOIP) call. Note that I have replaced actual phone numbers and domain names with *** for anonymity. I suspect perhaps a codec issue, but I haven't been able to identify the actual problem. Any ideas that will help me towards solving this problem is greatly appreciated. Regards, Helge [Feb 10 16:40:56] VERBOSE[5769] logger.c: -- event_offhook [Feb 10 16:40:56] VERBOSE[5769] logger.c: -- AST_STATE_DOWN: [Feb 10 16:40:56] VERBOSE[5769] logger.c: -- start mp_new [Feb 10 16:40:59] VERBOSE[5769] logger.c: -- event_dtmf # [Feb 10 16:40:59] DEBUG[5769] devicestate.c: Notification of state change to be queued on device/channel MP/1 [Feb 10 16:40:59] DEBUG[5767] devicestate.c: Changing state for MP/1 - state 0 (Unknown) [Feb 10 16:40:59] VERBOSE[5769] logger.c: -- event_dtmf * [Feb 10 16:40:59] VERBOSE[5769] logger.c: -- event_dtmf * [Feb 10 16:41:00] VERBOSE[5769] logger.c: -- event_dtmf * [Feb 10 16:41:00] VERBOSE[5769] logger.c: -- event_dtmf * [Feb 10 16:41:00] VERBOSE[5769] logger.c: -- event_dtmf * [Feb 10 16:41:00] VERBOSE[5769] logger.c: -- event_dtmf * [Feb 10 16:41:00] VERBOSE[5769] logger.c: -- event_dtmf * [Feb 10 16:41:01] VERBOSE[5769] logger.c: -- event_dtmf * [Feb 10 16:41:01] VERBOSE[5769] logger.c: -- event_dtmf * [Feb 10 16:41:01] VERBOSE[5769] logger.c: -- event_dtmf * [Feb 10 16:41:04] VERBOSE[5769] logger.c: -- event_digit_timer [Feb 10 16:41:04] VERBOSE[5769] logger.c: -- extension exists, starting PBX #** [Feb 10 16:41:04] DEBUG[5769] devicestate.c: Notification of state change to be queued on device/channel MP/1 [Feb 10 16:41:04] DEBUG[5767] devicestate.c: Changing state for MP/1 - state 0 (Unknown) [Feb 10 16:41:04] DEBUG[5901] pbx.c: Launching 'Dial' [Feb 10 16:41:04] VERBOSE[5901] logger.c: -- Executing [#**@default:1] Dial(MP/1, SIP/**@sipaccount|120|r) in new stack [Feb 10 16:41:04] DEBUG[5901] chan_sip.c: Asked to create a SIP channel with formats: 0x4 (ulaw) [Feb 10 16:41:04] DEBUG[5901] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) [Feb 10 16:41:04] DEBUG[5901] chan_sip.c: Setting NAT on RTP to On [Feb 10 16:41:04] DEBUG[5901] chan_sip.c: *** Our native formats are 0x2 (gsm) [Feb 10 16:41:04] DEBUG[5901] chan_sip.c: *** Joint capabilities are 0x0 (nothing) [Feb 10 16:41:04] DEBUG[5901] chan_sip.c: *** Our capabilities are 0xe (gsm|ulaw|alaw) [Feb 10 16:41:04] DEBUG[5901] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x2 (gsm) [Feb 10 16:41:04] DEBUG[5901] chan_sip.c: *** Our preferred formats from the incoming channel are 0x4 (ulaw) [Feb 10 16:41:04] DEBUG[5901] chan_sip.c: This channel will not be able to handle video. [Feb 10 16:41:04] DEBUG[5901] rtp.c: Channel 'MP/1' has no RTP, not doing anything [Feb 10 16:41:04] DEBUG[5901] channel.c: Not copying variable STACK-default-#**-1. [Feb 10 16:41:04] DEBUG[5901] chan_sip.c: Outgoing Call for ** [Feb 10 16:41:04] DEBUG[5901] chan_sip.c: Updating call counter for outgoing call [Feb 10 16:41:04] DEBUG[5901] chan_sip.c: Our T38 capability (0), joint T38 capability (0) [Feb 10 16:41:04] DEBUG[5901] chan_sip.c: ** Our capability: 0x6 (gsm|ulaw) Video flag: False [Feb 10 16:41:04] DEBUG[5901] chan_sip.c: ** Our prefcodec: 0x4 (ulaw) [Feb 10 16:41:04] VERBOSE[5901] logger.c: Audio is at 10.130.1.21 port 17800 [Feb 10 16:41:04] VERBOSE[5901] logger.c: Adding codec 0x4 (ulaw) to SDP [Feb 10 16:41:04] VERBOSE[5901] logger.c: Adding codec 0x2 (gsm) to SDP [Feb 10 16:41:04] VERBOSE[5901] logger.c: Adding non-codec 0x1 (telephone-event) to SDP [Feb 10 16:41:04] DEBUG[5901] chan_sip.c: -- Done with adding codecs to SDP [Feb 10 16:41:04]
Re: [asterisk-users] Light indicator managed by Asterisk
Eric Wieling wrote: pbx*CLI core show application minivmmwi Core show application minivmmwi core show function DEVICE_STATE Both of these must be a 1.6.x or newer, I have neither under 1.4 Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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] Light indicator managed by Asterisk
Check out http://kb.smartvox.co.uk/index.php/asterisk/sip-extensions/shared-voicemail-part2/ Date: Thu, 12 May 2011 14:38:46 -0400 From: supp...@drdos.info To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Light indicator managed by Asterisk Eric Wieling wrote: pbx*CLI core show application minivmmwi Core show application minivmmwi core show function DEVICE_STATE Both of these must be a 1.6.x or newer, I have neither under 1.4 Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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] Light indicator managed by Asterisk
Correct. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle Sent: Thursday, May 12, 2011 2:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Light indicator managed by Asterisk Eric Wieling wrote: pbx*CLI core show application minivmmwi Core show application minivmmwi core show function DEVICE_STATE Both of these must be a 1.6.x or newer, I have neither under 1.4 Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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] regarding core modules
Hi all, I would like to know what are core modules that are used for asterisk? can anyone help me regarding this... with regards, viswavardhan -- _ -- 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] Light indicator managed by Asterisk
You can, using device_state function (I use asterisk 1.6.2.X)
Here is a example for a conference... when sombody enter to conference a
light up on my aastra phone:
exten = s,1,Set(DEVICE_STATE(Custom:confer)=INUSE)
exten = s,n,Meetme(5000)
exten = s,n,Hangup
exten = h,1,MeetMeCount(5000,users)
exten = h,2,Gotoif($[${users} = 0]?end:noend)
exten = h,3(end),Set(DEVICE_STATE(Custom:confer)=NOT_INUSE)
exten = h,4(noend),Noop(Users number = ${users})
On your subscribe context:
exten = conf,hint,custom:confer
On the phone configuration, choice BLF and asign conf to the key
Regards
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Re: [asterisk-users] Realtime - ara180
Hi, look if you have res_config_mysql.so module instaled on your asterisk. On CentOS /usr/lib/asterisk/modules Regards -- _ -- 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] test call generator
dan, elder, I have played with scripts to generate calls and track their completion, email me off-list if you have questions. daveC Daniel - Asterisk wrote: Hello Everyone, I wonder if someone could share a manual about using SIPp for Asterisk's testing. I'll be gratefull Regards, Elder Arohuanca Lima - Peru On Tue, Sep 30, 2008 at 12:59 PM, zac wolfe zac.wo...@gmail.com mailto:zac.wo...@gmail.com wrote: Sipp looks pretty good! I don't know how I missed this one. This would've saved me tons of time a couple months ago. I plan on using it to load test using 2 Asterisk servers, one to initiate the SIP calls, the other to receive. Thanks for the tip Alex. Zac Wolfe Safi Systems LLC www.safisystems.com http://www.safisystems.com On Sat, Sep 27, 2008 at 5:58 AM, Alex Balashov abalas...@evaristesys.com mailto:abalas...@evaristesys.com wrote: What you are looking for is SIPP: http://sipp.sourceforge.net/ It won't intrinsically tell you anything about the data; it's up to you to appropriate the findings. But it accomplishes the generation of traffic (and dummy media!) on a technical level. Igor Hernandez wrote: Sam Tam wrote: Hello everyone I am trying to look for a free test call generator that will get me some stats like PDD, ASR and call quality etc on each route. As well as do test at every interval too If you know something like this please enlighten me. Sam ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hey Sam, I've been looking for such a tool also. I can't seem to find a tool that does those things. If nothing comes up in the next couple of weeks I'm going to code something up, I wouldn't mind letting you and anyone else who might be interested have the source once its done. Let me know if you find anything thats already out there in the meantime, might just save me a few hours of work. Regards, -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- SJREIA South Jersey Real Estate Investors Association Want to invest in Real Estate? come out and join over 450 real estate investors http://www.SJREIA.org Licensed NJ Real Estate Agent Buy This House REALTORs david.cant...@ibsonecall.com Mobile (856)813-7098 Office (856)324-4488 Pers Fax (646)827-7108 Ofc Fax (888)487-7711 Interlocking Business Solutions, LLC david.cant...@ibsonecall.com (856)581-8971 Home of the Videophone2009.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:
[asterisk-users] lead time for RPM's?
Hi all Usually I build asterisk from source, but recently have been doing a couple of test installations with packages from the Digium repository. About how long does it take to get from new release announcement into the Digium RPM repository? Specifically 1.8.4 CentOS hasn't made it to the rpm repository yet. Cassius -- _ -- 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] lead time for RPM's?
On 05/12/2011 02:40 PM, Cassius Smith wrote: Hi all Usually I build asterisk from source, but recently have been doing a couple of test installations with packages from the Digium repository. About how long does it take to get from new release announcement into the Digium RPM repository? Specifically 1.8.4 CentOS hasn't made it to the rpm repository yet. Cassius In most cases, we'll have RPMs built and available before the release notifications go out. However, we are currently in the process of rebuilding our build servers, so it has been delayed a few days. I expect that builds will be available in the next day or so. I'll make it a point to respond to this email when the new builds are available. -- _ -- 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] Light indicator managed by Asterisk
On 05/12/2011 07:12 PM, Carlos Chavez wrote: On Thu, 2011-05-12 at 18:50 +0200, Jonas Kellens wrote: Hello, is there some way to make Asterisk light up a certain light on an IP-phone ? Like MWI, the message waiting indicator can light up if there is voicemail. Could this light, or even other lights (like BLF-buttons) be used to give a visual notification to the user ? For example : if a certain value is set in the Mysql-DB and Asterisk reads out this value, can Asterisk react upon it inside the dialplan to make a light lit up ? 2nd example : if a certain extension is called, can we perform inside the dialplan an action that makes a light lit up on a Snom or Yealink IP-phone ? I don't know if all this is at all possible, but it doesn't harm asking I guess... If BLF works, then maybe more things are possible in the same way. Just thinking outside the box here. BLF lights can be manipulated with Hints and the DEVSTATE function to set custom device states. This way you can have a BLF light react to any event you want. Hello, I must say that I have succeeded in working with DEVSTATE to get a BLF-light in several colors. Which works great for what I want. Thank you for the feedback. Do you think it is also possible to get info displayed on the screen of the IP-phone ? Any idea how that would work ? Something tells me that this will depend on the brand/type of IP-phone. Kind regards, Jonas. -- _ -- 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 reload agents.conf ?
How to reload only agents.conf ? -- _ -- 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 reload agents.conf ?
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel Sent: Thursday, May 12, 2011 3:28 PM To: asterisk-users Subject: [asterisk-users] how to reload agents.conf ? How to reload only agents.conf ? [Danny Nicholas] Module reload chan_agent.so Should do the trick. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk 1.8 somehow dead
Guys! I am running 1.8 on production we have one PRI and 50 extensions. since last few days its working fine but today some how server load get high 194 % CPU and when i did asterisk -r i got CLI but no out put for any command. I check logs and nothing interesting there.. I am not using any advance feature just Voicemail, Meetme and calling.. Anybody having this kind of issue ? -S -- _ -- 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] When someone helps you, at least let them know if the problem is resolved or not
On 12/05/11 9:31 PM, Steve Totaro wrote: PS 42 is the answer, now what is the question. :) Heh, that might be one example where top posting would make sense ;-) -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/cc.php (Call Centre Solutions) -- _ -- 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] 1.8 and prematuremedia problem
hi: I report my issue as issue 19628. it is fixed and I run asterisk 1.8 in production now. thanks a lot for your help! Regards, tbskyd 2011/5/11 d tbsky tbs...@gmail.com: hi: ok I will create a bug report. and I found I still need prematuremedia=no in asterisk 1.6.2.18. yesterday I was testing at home with zoiper softphone + iax. today I test snom hardware sip phone and found that prematuremedia=no is still necessary. Regards, tbskyd 2011/5/11 satish patel satish...@hotmail.com: I am sorry about that but its interesting it doesn't work with 1.8 SVN I would say please report this bug so that way you can track issue, And may be in future it help us :) -S Date: Wed, 11 May 2011 01:31:34 +0800 Subject: Re: [asterisk-users] 1.8 and prematuremedia problem From: tbs...@gmail.com To: asterisk-users@lists.digium.com; satish...@hotmail.com hi: that issue is marked as fixed, so no more comment can be added :( anyway, I try the following combination: 1.8.3.2 + sig_pri patch 1.8 svn which already has sig_pri patched 1.8.4 + libpri patch (another unofficial patch in issue 18868) but none works. finally I downgrade to 1.6.2.18 and I found everything works. I don't even need to set prematuremedia with 1.6.2.18. so I think I will need to stay with 1.6.2 a little longer... thanks a lot for your help!! Regards, tbskyd 2011/5/10 satish patel satish...@hotmail.com: Also i would say add comment on following issue if after patch you having issue, That way it help community to fine tune patch. https://issues.asterisk.org/view.php?id=18868 Good luck From: satish...@hotmail.com To: tbs...@gmail.com Subject: Re: [asterisk-users] 1.8 and prematuremedia problem Date: Tue, 10 May 2011 07:43:47 -0400 CC: asterisk-users@lists.digium.com I have applied this patch in 1.8 svn branch and it works great for me. I have nothing special configuration just simple dial command for outgoing call. Also check there are progress=yes option in chan_dahdi -- Sent from my iPhone On May 10, 2011, at 5:58 AM, d tbsky tbs...@gmail.com wrote: hi: I apply sig_pri.c patch for 1.8.3.2 manually. (the patch can not apply to 1.8.3.2 or 1.8.4-rc3). but the situation is the same. do I need to play with other options with the patch? or I need newer asterisk versions to solve the problem? thanks a lot for information!! 2011/5/10 d tbsky tbs...@gmail.com: hi: thanks a lot for your quick reply. I saw that patch and think that it was already included in 1.8.3. now I know it will be included in 1.8.5. I will try it and thanks again for your kindly help!! 2011/5/10 Satish Patel satish...@hotmail.com: Apply this patch https://issues.asterisk.org/view.php?id=18868 -- Sent from my iPhone On May 9, 2011, at 9:57 PM, d tbsky tbs...@gmail.com wrote: hi: our current connection is below: sip phone---asteriskalcatel PBXPSTN asterisk and alcatel PBX is connected via E1 isdn-pri. when I use sip phone to dial outside PSTN world: 1. with 1.4 it is fine. 2. with 1.6.2, I need to set prematuremedia=no is sip.conf. or sip phone can not hear the ring and the beginning of the PSTN voice. 3. with 1.8.3.2, I can not hear ring and the beginning of the PSTN voice. I try to play options with prematuremedia and progressinband. but I can not find working settings. I don't know what other options I can try. thank a lot for information!! -- _ -- 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
Re: [asterisk-users] 1.8 and prematuremedia problem
hi: sorry. the issue number is 19268. not 19628. sorry about that!! Regards, tbskyd 2011/5/13 d tbsky tbs...@gmail.com: hi: I report my issue as issue 19628. it is fixed and I run asterisk 1.8 in production now. thanks a lot for your help! Regards, tbskyd 2011/5/11 d tbsky tbs...@gmail.com: hi: ok I will create a bug report. and I found I still need prematuremedia=no in asterisk 1.6.2.18. yesterday I was testing at home with zoiper softphone + iax. today I test snom hardware sip phone and found that prematuremedia=no is still necessary. Regards, tbskyd 2011/5/11 satish patel satish...@hotmail.com: I am sorry about that but its interesting it doesn't work with 1.8 SVN I would say please report this bug so that way you can track issue, And may be in future it help us :) -S Date: Wed, 11 May 2011 01:31:34 +0800 Subject: Re: [asterisk-users] 1.8 and prematuremedia problem From: tbs...@gmail.com To: asterisk-users@lists.digium.com; satish...@hotmail.com hi: that issue is marked as fixed, so no more comment can be added :( anyway, I try the following combination: 1.8.3.2 + sig_pri patch 1.8 svn which already has sig_pri patched 1.8.4 + libpri patch (another unofficial patch in issue 18868) but none works. finally I downgrade to 1.6.2.18 and I found everything works. I don't even need to set prematuremedia with 1.6.2.18. so I think I will need to stay with 1.6.2 a little longer... thanks a lot for your help!! Regards, tbskyd 2011/5/10 satish patel satish...@hotmail.com: Also i would say add comment on following issue if after patch you having issue, That way it help community to fine tune patch. https://issues.asterisk.org/view.php?id=18868 Good luck From: satish...@hotmail.com To: tbs...@gmail.com Subject: Re: [asterisk-users] 1.8 and prematuremedia problem Date: Tue, 10 May 2011 07:43:47 -0400 CC: asterisk-users@lists.digium.com I have applied this patch in 1.8 svn branch and it works great for me. I have nothing special configuration just simple dial command for outgoing call. Also check there are progress=yes option in chan_dahdi -- Sent from my iPhone On May 10, 2011, at 5:58 AM, d tbsky tbs...@gmail.com wrote: hi: I apply sig_pri.c patch for 1.8.3.2 manually. (the patch can not apply to 1.8.3.2 or 1.8.4-rc3). but the situation is the same. do I need to play with other options with the patch? or I need newer asterisk versions to solve the problem? thanks a lot for information!! 2011/5/10 d tbsky tbs...@gmail.com: hi: thanks a lot for your quick reply. I saw that patch and think that it was already included in 1.8.3. now I know it will be included in 1.8.5. I will try it and thanks again for your kindly help!! 2011/5/10 Satish Patel satish...@hotmail.com: Apply this patch https://issues.asterisk.org/view.php?id=18868 -- Sent from my iPhone On May 9, 2011, at 9:57 PM, d tbsky tbs...@gmail.com wrote: hi: our current connection is below: sip phone---asteriskalcatel PBXPSTN asterisk and alcatel PBX is connected via E1 isdn-pri. when I use sip phone to dial outside PSTN world: 1. with 1.4 it is fine. 2. with 1.6.2, I need to set prematuremedia=no is sip.conf. or sip phone can not hear the ring and the beginning of the PSTN voice. 3. with 1.8.3.2, I can not hear ring and the beginning of the PSTN voice. I try to play options with prematuremedia and progressinband. but I can not find working settings. I don't know what other options I can try. thank a lot for information!! -- _ -- 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] undefined symbol: cap_set_proc on several modules after installation from source
Hello Folks, What could be producing the following warnings on console, after an installation from source (Asterisk 1.4.41): [May 12 21:36:54] WARNING[15344]: loader.c:434 load_dynamic_module: Error loading module 'res_musiconhold.so': /usr/lib/asterisk/modules/res_musiconhold.so: undefined symbol: cap_set_proc [May 12 21:36:54] WARNING[15344]: loader.c:434 load_dynamic_module: Error loading module 'app_festival.so': /usr/lib/asterisk/modules/app_festival.so: undefined symbol: cap_set_proc [May 12 21:36:55] WARNING[15344]: loader.c:434 load_dynamic_module: Error loading module 'app_ices.so': /usr/lib/asterisk/modules/app_ices.so: undefined symbol: cap_set_proc [May 12 21:36:55] WARNING[15344]: loader.c:434 load_dynamic_module: Error loading module 'app_mp3.so': /usr/lib/asterisk/modules/app_mp3.so: undefined symbol: cap_set_proc [May 12 21:36:55] WARNING[15344]: loader.c:434 load_dynamic_module: Error loading module 'app_nbscat.so': /usr/lib/asterisk/modules/app_nbscat.so: undefined symbol: cap_set_proc [May 12 21:36:55] WARNING[15344]: loader.c:434 load_dynamic_module: Error loading module 'app_externalivr.so': /usr/lib/asterisk/modules/app_externalivr.so: undefined symbol: cap_set_proc [May 12 21:36:55] WARNING[15344]: loader.c:434 load_dynamic_module: Error loading module 'app_dahdiras.so': /usr/lib/asterisk/modules/app_dahdiras.so: undefined symbol: cap_set_proc Uname -a: Linux eslackware 2.6.33.4-smp #2 SMP Wed May 12 22:47:36 CDT 2010 i686 Intel(R) Core(TM)2 Duo CPU E4500 @ 2.20GHz GenuineIntel GNU/Linux gcc -v: Reading specs from /usr/lib/gcc/i486-slackware-linux/4.4.4/specs Target: i486-slackware-linux Configured with: ../gcc-4.4.4/configure --prefix=/usr --libdir=/usr/lib --enable-shared --enable-bootstrap --enable-languages=ada,c,c++,fortran,java,objc --enable-threads=posix --enable-checking=release --with-system-zlib --with-python-dir=/lib/python2.6/site-packages --disable-libunwind-exceptions --enable-__cxa_atexit --enable-libssp --with-gnu-ld --verbose --with-arch=i486 --target=i486-slackware-linux --build=i486-slackware-linux --host=i486-slackware-linux Thread model: posix gcc version 4.4.4 (GCC) cat /etc/slackware-version: Slackware 13.1.0 -- Jose P. Espinal http://www.eslackware.com IRC: [OFTC|FreeNode] Khratos @ #slackware | #asterisk/-doc/-bugs -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk 1.8 + google voice
somewhere along the way, i noticed incoming calls from google voice are no longer working on my asterisk 1.8.3.2 system. When the call comes in, asterisk immediately prints on the console: == Spawn extension (google-in, s, 2) exited non-zero on 'Gtalk/+12153930924-f947' [May 12 22:47:18] NOTICE[13835]: chan_gtalk.c:1977 gtalk_parser: Remote peer reported an error, trying to establish the call anyway the calling side just hears ringing. i have plenty of debug info, but nothing too interesting. anyone else having this problem ? or is it time for bug report ? -- Jeremy Kister http://jeremy.kister.net./ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk 1.8 + google voice
On 5/12/2011 11:08 PM, Jeremy Kister wrote: [May 12 22:47:18] NOTICE[13835]: chan_gtalk.c:1977 gtalk_parser: Remote peer reported an error, trying to establish the call anyway I found the problem, and I am sending in a bug report :) if anyone is interested, the issue is 19286 (i'll be completing it shortly) -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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] 1.8.4 Core Dump after installing from source
Hello, After installing Asterisk from source in Slackware 13.1, I get the following error: Error loading module 'res_config_odbc.so': /usr/lib/asterisk/modules/res_config_odbc.so: undefined symbol: ast_odbc_clear_cache Then a core dump. If I change the /etc/asterisk/modules.conf in order to preload the 'res_odbc.so' module, then the error dissapears, *but* still crashes with core dump Could someone point me out as where to start looking, or point me out to some documenation? Aditional Info: I enabled ODBC voicemail storage through the command line with: make menuselect.makeopts menuselect/menuselect --disable-category MENUSELECT_OPTS_app_voicemail menuselect.makeopts menuselect/menuselect --enable ODBC_STORAGE menuselect.makeopts Could it be that ODBC_STORAGE is causing problems with FILE_STORAGE, even if I explicitly disabled FILE_STORAGE? I also used 'strip' on the binaries (could that be the reason?): find . | xargs file | grep executable | grep ELF | cut -f 1 -d : | xargs strip --strip-unneeded 2 /dev/null find . | xargs file | grep shared object | grep ELF | cut -f 1 -d : | xargs strip --strip-unneeded 2 /dev/null Regards, -- Jose P. Espinal http://www.eslackware.com IRC: [OFTC|FreeNode] Khratos @ #slackware | #asterisk/-doc/-bugs -- _ -- 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
