[asterisk-users] How to put your call on hold with asterisk Dialplan
Hi List, Is it possible to put your call on hold after bridge with another call ? I want to use asterisk dialplan not softphone button or IP-phone device. If yes then how to get back on bridge mode of on-hold call again ? - 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] Skype for Asterisk - RIP
On Wednesday 25 May 2011, randulo wrote: On Tue, May 24, 2011 at 10:50 PM, Matt Darnell mattdarn...@gmail.com wrote: We expect that users of Skype for Asterisk will be able to continue using their Asterisk systems on the Skype network until at least July 26, 2013. Skype may extend this at their discretion. It's widely believed. However, it's very possible that this was not a Microsoft decision but planned by Skype before the acquisition. Bgeh. Serves 'em right for using that POC! Who honestly *hadn't* seen this coming since the day Skype was first released? A subscriber with one telephone company has to be able at least to contact subscribers with any other telephone company -- that much ought to be self-evident. It is also highly desirable, where multiple telephone companies are competing for business in the same physical space, to be able to use the same equipment with any of them. Such interoperability requires open standards that can be implemented by anybody, and the most preferable way to achieve this is through an Open Source reference implementation (not just Asterisk; think OpenBSD and the Secure Shell, or Apache and HTTP). Skype's secretive, proprietary nature -- surely the absolute antithesis of what telecommunications needs to be about -- means that only Skype subscribers can talk to other Skype subscribers. (It also potentially runs afoul of some European countries' telecommunications deregulation and competition laws -- except, as we all know, normal laws don't apply anywhere there is a computer involved). We in the Asterisk user community should be persuading Skype users to move to proper VoIP solutions, sooner rather than later -- even if that means recommending another proprietary product as a pragmatic intermediate measure. At least Caged products which *correctly* implement Open standards are likely candidates for Free drop-in replacements later. (Paraphrased and expanded from an earlier post by me on another forum.) -- 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] Skype for Asterisk - RIP
On Wed, May 25, 2011 at 10:53 AM, A J Stiles asterisk_l...@earthshod.co.uk wrote: Bgeh. Serves 'em right for using that POC! Who honestly *hadn't* seen this coming since the day Skype was first released? Tim Panton, who's beenworking with SfA since it came out, posted this article today: http://vuc.li/meBRJd /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] synway
hi: yes, Synway's asterisk cards work with zaptel and dahdi. it is a good product. Best regards, James.zhu Doing asterisk/PRI/ss7/dahdi, linux, asterisk cards, gateway(fxs/fxo/pri-SIP). website: www.voipviews.com From: l...@lopl.net Date: Wed, 25 May 2011 09:15:47 +0430 To: asterisk-users@lists.digium.com Subject: [asterisk-users] synway Dear,do you have any successful experience for installing SHT-8C/PCI/FAX (synway) with asterisk ?is it compatibe with asterisk (dahdi/zaptel)? best -- Pezhman Lali -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk cluster and call pickup
I'm planning to migrate 500+ extensions from a legacy PBX to Asterisk. Some will be SIP, some will be DAHDI FXS. I want to deploy a load-balancing cluster using DUNDi with regcontext so all servers will know where to find all extensions. DAHDI extensions will have their dedicated server, SIP extensions will be spread out on 3 different servers. Now I'm wondering how would call pickup work on that scenario. Is it possible (maybe with distributed devstates) to pickup a ringing channel on a different server? Or is there a different way to load balance while keeping call pickup functionality intact? Any input on this is very welcome. -- _ -- 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] AJAM XML output not valid xml
On Mon, 2011-05-23 at 15:41 +0100, Ishfaq Malik wrote: Hi I'm using asterisk 1.8.3.2 and have been implementing AJAM. I've noticed the final '' is missing from every response I've had so far. Here is an example ajax-response response type='object' id='unknown'generic response='Success' message='Authentication accepted' //response /ajax-response Has anyone else noticed this? Is it a bug in the code or possibly a config setting I've missed? Thanks Ish Can someone else please verify that this is happening to them as well and if so I'll raise the issue... -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- 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 hint SIP presence
On Tue, May 24, 2011 at 10:26 PM, Deka, Rajib IN MAA SL rajib.d...@siemens.com wrote: Hello List, Asterisk CLI command “core show hints” gives the list of hint extension configured and its presence status. In command output there is a field called “watchers” and it contains a numeric value of number of subscriptions’ registered for that particular extension. So, is there any CLI command to check who the watchers for an extension are? I use 'sip show subscriptions' to see what peers watch which hint. -- _ -- 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] anyone who has experience with chinese clone cards like zycoo, ctvon, chinaroby, etross, iit, realtone
Asim Amin asim at horizontech.biz writes: Hello All,Anyone who has experience using Digium analog card clones from any of the following:1. Zycoo2. CTVON3. Chinaroby4. Etross5. Immediate IT (IIT)6. Realtone and can give review which one is good quality with easy configuration and error free running. Also since some of these manufacture only analog cards, does anyone have any experience using these in a single system with digital cards from other manufacturers like Openvox? -- Asim AminPartnerTechnical Manager, Telco DivisionHorizon TechnologiesCell: +92-323-3314151E-mail: asim at horizontech.bizWeb: http://horizontech.biz http://hostht.com -- _ I don’t have experience with the rest, except Chinaroby. I bought two cards, (TDM410P TE405P) from eBay. I have both bad experience!! I run the latest dahdi 2.4.1 The LEDs on the TDM410P do not lite. Occasionally, the FXS ports will generate lots of noise. The FXS modules indeed feel very hot. It was a nightmare to install the TE405 as well. It was not recognized by Elastix 2.0 If I use Elastix 1.6, the ports appear. However, I cannot make outgoing calls. I finally downgrade to Elastix 1.5 on Dahdi 2.3. The outgoing/incoming calls work but there are some “clicking” noise in the first about 10s. My client eventually refuses to accept this system. I have to replace the card from another supplier. No more clicking noise. So far, it works well for the past 4 months. -- _ -- 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 + Patton ISDN2e gateway (UK)
2011/5/24 Gordon Henderson gordon+aster...@drogon.net I know I asked this some time back, and I got no response then, and neither did someone who asked at the start of 2009 either by the looks of it (other than a reply from me to use a PCI card!!!) However I now have a client who'd bought one of these boxes and installed ISDN2e - against my advice (because I got no feedback here, therefore they're junk with asterisk, right?) However I'm currently stuck with it and it's a right royal PITA to program up by the look of it - I've done all the obvious stuff, and am now wading through their 600+page manual, but ugh... So if anyone has used one with asterisk as a bridge to ISDN2e in the UK, would you mind sharing a config with me? I have it registering (it pretends to be a SIP phone, it registers with Asterisk), but when I try to send a call to it, it times out - I do see debug on it's console though - same when I send a call into it from the PSTN side - I see ISDN debug messages, but no SIP activity. And I thought Mediatrix units were bad - but at least they are relatively easy to program... Bah. Any clues appreciated, Hi, I don't have any specific knowledge of ISDN2e in the UK, but I can configure patton devices (through embedded web server) to work with Asterisk. Do not hesitate to contact me off-list if I can be of any help. Regards Thanks, 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 -- _ -- 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 multiple queue
Hey Guys! We had migrate asterisk 1.2 to 1.8 now big issue is queue system. Before we had 3 queues and we were using AgentCallbackLogin but now its quite difficult to use AddQueueMember. Is there any easy way to logged into multiple queue using AddQueueMember ? and restrict agent for specific queue ? -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
[asterisk-users] Tr : how to user SIP realtime option
Hi Guys, Has anyone been able to configure the realtime using Asterisk 1.4.26.1? So far I've successfully configured the realtime using Asterisk 1.6.2 but while trying to use 1.4, Asterisk keep restarting. Any clue or suggestion?-- _ -- 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] Tr : how to user SIP realtime option
I have configured tons of 1.4.x installs with various levels of realtime integration. Are you asking about the specific sub-version 1.4.26.1, or the 1.4 branch in general? Also, in order for anyone to help you, we'd need some info like how ou set up realtime, log outputs, etc Sent from my iPhone On May 25, 2011, at 4:33 PM, Dr Ox dr_o...@yahoo.fr wrote: Hi Guys, Has anyone been able to configure the realtime using Asterisk 1.4.26.1? So far I've successfully configured the realtime using Asterisk 1.6.2 but while trying to use 1.4, Asterisk keep restarting. Any clue or suggestion? -- _ -- 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] Tr : how to user SIP realtime option
On 5/25/2011 4:33 PM, Dr Ox wrote: Hi Guys, Has anyone been able to configure the realtime using Asterisk 1.4.26.1? So far I've successfully configured the realtime using Asterisk 1.6.2 but while trying to use 1.4, Asterisk keep restarting. Any clue or suggestion? By the way, I forgot to add, I'd be more than happy to help you with your SIP realtime integration. Please post the relevant configuration files (such as extconfig.conf, odbc.ini odbc-inst.ini, etc) and a mysqldump of structure and data from your sip table(s). Full log output from your server would also be helpful. You may reply to me offlist if you 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] Asterisk 1..8 multiple queue
On 5/25/2011 12:32 PM, satish patel wrote: Hey Guys! We had migrate asterisk 1.2 to 1.8 now big issue is queue system. Before we had 3 queues and we were using AgentCallbackLogin but now its quite difficult to use AddQueueMember. Is there any easy way to logged into multiple queue using AddQueueMember ? and restrict agent for specific queue ? -S Use of the realtime architecture for queue members is my preferred method. -- _ -- 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 multiple queue
Thanks for reply but is there any alternative way? Because we don't have mysql and we dont want to use mysql. -- Sent from my iPhone On May 25, 2011, at 6:43 PM, Sherwood McGowan sherwood.mcgo...@gmail.com wrote: On 5/25/2011 12:32 PM, satish patel wrote: Hey Guys! We had migrate asterisk 1.2 to 1.8 now big issue is queue system. Before we had 3 queues and we were using AgentCallbackLogin but now its quite difficult to use AddQueueMember. Is there any easy way to logged into multiple queue using AddQueueMember ? and restrict agent for specific queue ? -S Use of the realtime architecture for queue members is my preferred method. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1..8 multiple queue
On 05/25/2011 08:20 PM, Satish Patel wrote: Thanks for reply but is there any alternative way? Because we don't have mysql and we dont want to use mysql. Who said you have to use MySQL for RT? -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.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] Asterisk 1..8 multiple queue
Yes, there are other ways, I was only offering the solution that has worked best for me. Keep in mind, you are not limited to MySQL for realtime, Asterisk can use any ODBC DSN for the data backend. Oracle, Access, MSSQL are all examples, if I recall correctly you can even connect SQLite and DB2. However, let me ask you this...what trouble are you having with AddQueueMember and it's related applications that is making it hard for you? Sent from my iPhone On May 25, 2011, at 7:20 PM, Satish Patel satish...@hotmail.com wrote: Thanks for reply but is there any alternative way? Because we don't have mysql and we dont want to use mysql. -- Sent from my iPhone On May 25, 2011, at 6:43 PM, Sherwood McGowan sherwood.mcgo...@gmail.com wrote: On 5/25/2011 12:32 PM, satish patel wrote: Hey Guys! We had migrate asterisk 1.2 to 1.8 now big issue is queue system. Before we had 3 queues and we were using AgentCallbackLogin but now its quite difficult to use AddQueueMember. Is there any easy way to logged into multiple queue using AddQueueMember ? and restrict agent for specific queue ? -S Use of the realtime architecture for queue members is my preferred method. -- _ -- 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
