Re: [asterisk-users] Multiple IAX2 Trunks Load balancing
yeah -- searching how to perform this magic ... On Fri, Dec 13, 2013 at 2:29 PM, Steven Howes steve-li...@geekinter.netwrote: On 13 Dec 2013, at 07:48, Muhammad Usman replyus...@gmail.com wrote: Hi - I have 2 Asterisk servers connected using 05 IAX2 trunks. I want to load balance incoming calls over IAX2 trunks. If any trunk goes down the calls traffic will be shared with other available trunks. When it gets Up the script is supposed to perform as desired i.e in load balance mode. Sounds wonderful. 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 -- Regards: (Muhammad υѕмαη ) -- _ -- 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 IAX2 Trunks Load balancing
Friends let me define the scenario please; Scenario: 2 asterisk servers (A B) are connected using 05 IAX2 trunks between them. The machine A is running asterisk Openvpn server in TUN mode (5 instances with difference IP addresses for clients). The machine B is running asterisk with 05 OpenVPN clients using 05 bandwidths. The IAX trunks are established between each pair of P-2-P ip address of machine A (The OPENVPN Server) machine B (The Openvpn client). Requirement: Required dial plan configuration at machine A for incoming calls from VoIP Switch/VOS which can forward the calls to IAX2 trunks in round robin fashion like Load Balancing. If any trunk goes down it starts forwarding the traffic to other available trunks when it gets UP the dialplan should perform as desired. Like L.B Fail-over scenarios. On Fri, Dec 13, 2013 at 8:52 PM, Hans Witvliet aster...@a-domani.nl wrote: On Fri, 2013-12-13 at 06:20 -0600, Don Kelly wrote: On Fri, 2013-12-13 at 12:48 +0500, Muhammad Usman wrote: Hi - I have 2 Asterisk servers connected using 05 IAX2 trunks. I want to load balance incoming calls over IAX2 trunks. If any trunk goes down the calls traffic will be shared with other available trunks. When it gets Up the script is supposed to perform as desired i.e in load balance mode. Thanks in advance. Hans said: Perhaps it is possible to do the L.B. at the O.S. or network level, and let all trunks appear to asterisk to one single trunk. Don asks: What's the value of load balancing multiple IAX trunks between the same system pair? What resources are being balanced? ++ Perhaps the O.P. can explain about his intentions... In some situations it makes sense though: If you have to connect two servers, and use different kind of infrastructure / multiple providers... hw -- _ -- 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 -- Regards: (Muhammad υѕмαη ) -- _ -- 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 IAX2 Trunks Load balancing
Hi - I have 2 Asterisk servers connected using 05 IAX2 trunks. I want to load balance incoming calls over IAX2 trunks. If any trunk goes down the calls traffic will be shared with other available trunks. When it gets Up the script is supposed to perform as desired i.e in load balance mode. Thanks in advance. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
you running GSM FWTs with asterisk ? On Mon, Apr 25, 2011 at 6:51 AM, Abid Saleem abid_aster...@hotmail.comwrote: HI, I am trying to setup a Class 4 termination setup using a kind of channel hunting scenerio. I have some SIP DID numbers assigned from the local telecom provider for termination. MY call comes from my wholesale client and lands on a switch, then it is routed to asterisk. I want asterisk to route this call to my local DID provider on the next available channel with DID number as the new Caller ID. This is just like GSM gateway that recieves the call and then re-originates the call using the next available SIM card number. Can someone help me how can I configure Asterisk to perform this? Thanks Abid. -- _ -- 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 -- Regards: (Muhammad υѕмαη ) -- _ -- 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] Fix Fake Answer Supervision In asterisk1.6
Hi, I have installed asterisk1.6+DAHDI for TDM2400P Digium card. When call hits the box, the gets answered even the other end phone in not picked. How can I fix this as ideally it should answer the call when other end phone is picked. -- _ -- 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] digim tdm2400p fxo fake answer supervision problem.
Hi. I am using Digium TDM2400PFXO with Asterisk1.6. When call sets in the box , it answers the call even the phone is not picked. ideally it should answer the call when the phone is picked up. Its charging the clients. Please let me know how can I cover this ? Thanks in advance. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Snom870 sidecar
Hi Olivier, General Availability for snom8xx sidecar: ~March 2010 UT -Ursprüngliche Nachricht- Von: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Im Auftrag von asterisk-users-requ...@lists.digium.com Gesendet: 19 October 2009 15:15 An: asterisk-users@lists.digium.com Betreff: asterisk-users Digest, Vol 63, Issue 49 Message: 3 Date: Mon, 19 Oct 2009 08:16:00 +0200 From: Olivier oza-4...@myamail.com Subject: Re: [asterisk-users] Snom870 sidecar To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: 442fbb120910182316t74062a9di4685e7d746a8e...@mail.gmail.com Content-Type: text/plain; charset=windows-1252 2009/10/18 Christian Stredicke christian.stredi...@snom.de The sidecar is not in the market yet. Any targeted schedule ? Just some information? It has its own CPU, Ethernet port and it is able to run applications (for example, Asterisk). Very interesting ! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] Snom Phones Registration/Failover Feature
Hi Raimund, snom uses basically the same concept. As explained under: http://wiki.snom.com/Settings/user_failover_identity. You select the line id that should be used when a registration fails. Regards, Usman -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of asterisk-users-requ...@lists.digium.com Sent: Thursday, August 13, 2009 4:46 PM To: asterisk-users@lists.digium.com Subject: asterisk-users Digest, Vol 61, Issue 34 Send asterisk-users mailing list submissions to asterisk-users@lists.digium.com Date: Thu, 13 Aug 2009 16:10:15 +0200 From: Raimund Sacherer r...@runsolutions.com Subject: [asterisk-users] Snom Phones Registration/Failover Feature To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: 2760e83d-c16f-43d6-b89d-9fac2be55...@runsolutions.com Content-Type: text/plain; charset=US-ASCII; format=flowed; delsp=yes Hello Mailinglist, i was reading a paper regarding a Asterisk clustering solution and they where pretty excited about a feature in polycom phones: You can add a registration to a primary asterisk server You can add a registration to a secondary asterisk server The polycom phones will talk to the primary server as long as all goes well, If they have a problem with an INVITE, they automatically register to the secondary asterisk server and start using them. Every few seconds (I think it was 30) the phone tries again to register on the primary server, if this succeeds, it uses the primary again. This is in my oppinion a pretty decent way of doing failover (reminds me of radius). It beats using Heartbeat and IP Takeover and all the hassle you (could) have with this solution. I was reading in the documentation about the SNOM phones (mainly 300) but I did not find anything in the users-pdf's or on there knowledgebase/website which would tell me if this is possible, there is something for failover configuration but it is not explained at all. It's highly appreciated if someone with insight could explain to me or point me to the right documentation on how/if this works with SNOM's. Thank you, best regards -- Raimund Sacherer - RunSolutions Open Source It Consulting - Parc Bit - Centro Empresarial Son Espanyol Edificio Estel - Local 3D 07121 - Palma de Mallorca Baleares Virus checked by G DATA AntiVirus Version: AVF 19.497 dated 13.08.2009 Virus news: www.antiviruslab.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
Re: [asterisk-users] SNOM on Do Not Call list????
Hi, Unfortunately that is true for the time being. Since we moved our main office to new premises, our telecom provider has failed setup services in time. Forums and otrs is online and we hope to have the phones working ASAP. We appreciate your understanding. Regards, Usman. -- Usman Tahir snom technology AG http://www.snom.com -Original Message- Date: Thu, 13 Mar 2008 10:28:29 -0400 From: Drew Gibson [EMAIL PROTECTED] Subject: [asterisk-users] SNOM on Do Not Call list To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1; format=flowed Some light relief SNOM say Please note that you will not be able to reach us by phone. http://www.theregister.co.uk/2008/03/13/dont_call_us/ regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calls Being Randomly Bridged
Hi Mike, For starters disable Call join on Xfer (2 calls): on the phones. Since the setup has 6.2.x, it most likely doesn't have the setting Allow incoming calls redirection through programmable keys available on 7.1.30 for snom360. You might wanna try this version on a test system and see if it helps in that environment. The problem, as discussed, seems to be originating when calls are parked on orbits that are mixing the two calls together. As long as you are debugging the issue, you should probably ask your friend to disable this practice and have a look at the call parking mechanism. Regards, Usman. - Usman Tahir snom technology AG www.snom.com This e-mail may contain confidential and/or privileged information. If you are not the intended recipient (or have received this e-mail in error) please notify the sender immediately and destroy this e-mail. Any unauthorized copying, disclosure or distribution of the material in this e-mail is strictly forbidden. Diese E-Mail könnte vertrauliche und/oder rechtlich geschützte Informationen enthalten. Wenn Sie nicht der richtige Adressat sind oder diese E-Mail irrtümlich erhalten haben, informieren Sie bitte sofort den Absender und vernichten Sie diese Mail. Das unerlaubte Kopieren sowie die unbefugte Weitergabe dieser Mail sind nicht gestattet. - -Original Message- Message: 11 Date: Sat, 19 Jan 2008 21:32:42 -0500 From: Michael J. Liberatore [EMAIL PROTECTED] Subject: [asterisk-users] Calls Being Randomly Bridged To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Hi i have a friend who i setup an asterisk system for at his doctors office. it has 3 snom 360 phones with 6.2.x stable firmware and latest asterisk 1.4 and zaptel. They have the digium 4 port fxo card. They are extremely upset because calls are being randomly bridged for no rhyme or reason. They say that callers will call in and sometimes get connected with other callers, or they will be in the queue and then be talking to another caller waiting in the queue or on hold. Or they will be talking to a patient and then have another patient end up on the conversation. They are freaking out because of hippa and laws that govern privacy but i have no clue why. I assume most cases are conference calls being initiated by accident. So any help would be greaat. maybe just disabling conference calls would be a good start but i dont know how with sip phones. or maybe this is a bug? unfortuinately they dont give me much info and i dont use the phones so i dont have any specific logs to show, they just call me freaking out saying this stuff but they rarely can give me a specific call cause they get so many. thanks mike ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RE: Snom has dialtone after putting a person on hold
Hi Ron, You can change this setting through the web interface Advanced/Audio/Dialtone during Hold. Hope that helps! Regards, Usman. - Usman Tahir snom technology AG Gradestraße 46 www.snom.com This e-mail may contain confidential and/or privileged information. If you are not the intended recipient (or have received this e-mail in error) please notify the sender immediately and destroy this e-mail. Any unauthorized copying, disclosure or distribution of the material in this e-mail is strictly forbidden. Diese E-Mail könnte vertrauliche und/oder rechtlich geschützte Informationen enthalten. Wenn Sie nicht der richtige Adressat sind oder diese E-Mail irrtümlich erhalten haben, informieren Sie bitte sofort den Absender und vernichten Sie diese Mail. Das unerlaubte Kopieren sowie die unbefugte Weitergabe dieser Mail sind nicht gestattet. - -- Message: 15 Date: Thu, 18 Jan 2007 11:10:47 -0700 From: Ron McCarthy [EMAIL PROTECTED] Subject: [asterisk-users] Snom has dialtone after putting a person on hold To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 Hi List, I cant seem to find the setting that changes this! You put a person on hold, they are on hold like normal, but after a few seconds the phone will then start having dialtone coming from the speakerphone, really weird!! Anyone know how to fix this? I see where it could be nice, but in this case, we just want them on hold is all, no dialtone! Any help would be great! Thanks! Ron ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Snom 360 doesn't register after reboot
Hi Domenico, Try Ver. 6.2.1. This problem is fixed in it. http://www.snom.com/wiki/index.php/Beta_Firmware#Release_6.2.1 Regards, Usman Tahir snom technology AG -- Message: 17 Date: Tue, 20 Jun 2006 18:18:43 +0200 From: Mimmus [EMAIL PROTECTED] Subject: [Asterisk-Users] Snom 360 doesn't register after reboot To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Hi, I'm trying my new Snom 360 phone (6.2 firmware) and I'm seeing that it doesn't register with the Asterisk 1.2.9.1 server after a reboot. I need to click Re-register in the web interface. I set: - Support broken Registrar: On - RTP Encryption: Off Any help? -- Domenico Viggiani ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Snom 360 problems
Detailed info about snom beta firmware can also be found at snom-wiki e.g. http://snom.com/wiki/index.php/Beta_Firmware#Release_Notes Regards, - Usman Tahir snom technology AG - Date: Sat, 25 Mar 2006 11:53:24 -0800 (PST) From: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] RE: Snom 360 problems To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: TEXT/PLAIN; charset=US-ASCII; format=flowed On Fri, 24 Mar 2006, Usman Tahir wrote: For the conf on Xfer issue, use the latest beta http://fox.snom.com/download/snom360-5.5.1b-beta-SIP-j.bin what's the changelog for 5.5.1b? -Dan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Snom 360 problems
Release Notes for recent snom360 beta firmware: Release 5.5.1: o GUI: fixed consultative Xfer with fkeys Release 5.5: o GUI: fixed cursor handling (scrolling, backspace) in edit number state o GUI: put last active call on hold on top in holding/transfer Release 5.4: o GUI: added shared line LED blink when holding o SIP: fixed bug in ENUM lookup o LID: fixed port access for keep_alive where it could access a port that didn't exist anymore Release 5.3.6: o LID: made sure audio channels are off in idle mode under all scenarios Release 5.3.5: o GUI: added cwi ringer indication o GUI: fixed unnecessary dialog state switches on shared line offhook o GUI: status led for missed calls o SIP: RAck in PRACK was buggy o SIP: added call pickup for shared lines Release 5.3.4: o SIP: added +sip.rendering parameter for BLA hold/resume NOTIFYs o SIP: NOTIFYs with subscription-state: terminated remove the subscription Release 5.3.3: o GUI: fixed DND o GUI: fixed bug in displaying old voice mail messages o SIP: display local LED status for shared lines o WEB: + in settings value isn't anymore replaced by its hex value on settings dump web interface page o WEB: further enhanced french translation o SRTP: fixed bug with auto-answer Release 5.3.2: o GUI: setting_server can be set manually via GUI menu (snom360) o GUI: ringer device should not switch to speaker if headset is enabled o GUI: dkeys (e.g. Redial, Retrieve) are working in edit number state, too o SETTINGS: if setting_server is IP:port only, make a valid URL out of it o SIP: display local LED status for shared lines o SRTP: fix bug with auto-answer Release 5.3.1: o GUI: Shared Lines can be mapped to LEDs o LID: random number generated from random audio data Release 5.3: o GUI: blind-xfer via programmable keys doesnt require pressing the Enter key o GUI: incoming call context can be switched with the cursor o GUI: fixed freezing during calls on hold o GUI: added setting cancel_on_hold which, if set to false, makes the phone ignore any cancel key press in holding state o GUI: fixed DND, wasn't working properly after reboot during DND on o GUI: enhanced french translation o GUI: fixed, mute key stops working after 20 seconds if no DNS server is reachable o LID: further reduced ringer volumes o SIP: unsupported p-time values for codecs in responses disconnects the call o SIP: treat all return codes 100 and 180 as 180 Ringing o WEB: enhanced french translation - Usman Tahir snom technology AG -- Message: 13 Date: Sat, 25 Mar 2006 11:53:24 -0800 (PST) From: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] RE: Snom 360 problems To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: TEXT/PLAIN; charset=US-ASCII; format=flowed On Fri, 24 Mar 2006, Usman Tahir wrote: For the conf on Xfer issue, use the latest beta http://fox.snom.com/download/snom360-5.5.1b-beta-SIP-j.bin what's the changelog for 5.5.1b? -Dan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Snom 360 problems
Hi Brian, For the conf on Xfer issue, use the latest beta http://fox.snom.com/download/snom360-5.5.1b-beta-SIP-j.bin Regards, - Usman Tahir snom technology AG Gradestraße 46 D-12347 Berlin. Tel: +49 30 398330 Fax: +49 30 39833111 [EMAIL PROTECTED] www.snom.com This e-mail may contain confidential and/or privileged information. If you are not the intended recipient (or have received this e-mail in error) please notify the sender immediately and destroy this e-mail. Any unauthorized copying, disclosure or distribution of the material in this e-mail is strictly forbidden. Diese E-Mail könnte vertrauliche und/oder rechtlich geschützte Informationen enthalten. Wenn Sie nicht der richtige Adressat sind oder diese E-Mail irrtümlich erhalten haben, informieren Sie bitte sofort den Absender und vernichten Sie diese Mail. Das unerlaubte Kopieren sowie die unbefugte Weitergabe dieser Mail sind nicht gestattet. - Date: Fri, 24 Mar 2006 12:41:26 -0500 From: Brian Kennedy [EMAIL PROTECTED] Subject: [Asterisk-Users] Snom 360 problems To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1; format=flowed Anyone have a Snom they're happy with? How did you manage that? :) I have a system of: Asterisk 1.2.3 2 Wildcard TDM400P Rev I and E/F 1 Snom 360 + sidecar ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] snom Firmware 5.0.
Old Ringer 2 4 will be available as 9 10 (in addition to the existing melodies) in Version 5.1 to be released in a few days. Its better than wasting bandwidth downloading such a custom melody, as Ringer2 seems so popular. Hope that will suffice... Regards, Usman. Message: 13 Date: Tue, 3 Jan 2006 10:05:35 -0600 From: Joe Pukepail [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] snom Firmware 5.0. To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 I agree, I liked the old ringtone 2 also (just a beep), I use it at my desk, If I'm there I can pick it up and it wasn't obnoxious enough to disturb others. Please email it to me if you get it in the format needed. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] snom Firmware 5.0.
Hi Remco, Old Ringer 2 is not there on the phone anymore, perhaps you can use another ring melody or a suitable custom melody: The wav file itself should be a PCM encoded 8 KHz file at 16bit mono. The time for loading the file should not be longer then 3 seconds ! And the size should be below 250KB. To create this format from mp3: /usr/bin/mpg123 -m -r 8000 -w - -n 190 -q test.mp3 test.wav To convert an existing WAV file: sox GENERIC.wav -c 1 -r 8000 -w SNOM.wav * The -c 1 flag makes the output mono. * The -r 8000 flag makes the output a 8kHz sample. * The -w flag uses 16 bits (word) per sample. Regards, Usman. - Usman Tahir snom technology AG Gradestraße 46 D-12347 Berlin. Tel: +49 30 398330 Fax: +49 30 39833111 [EMAIL PROTECTED] www.snom.com This e-mail may contain confidential and/or privileged information. If you are not the intended recipient (or have received this e-mail in error) please notify the sender immediately and destroy this e-mail. Any unauthorized copying, disclosure or distribution of the material in this e-mail is strictly forbidden. Diese E-Mail könnte vertrauliche und/oder rechtlich geschützte Informationen enthalten. Wenn Sie nicht der richtige Adressat sind oder diese E-Mail irrtümlich erhalten haben, informieren Sie bitte sofort den Absender und vernichten Sie diese Mail. Das unerlaubte Kopieren sowie die unbefugte Weitergabe dieser Mail sind nicht gestattet. - -Original Message- From: Remco Barende [mailto:[EMAIL PROTECTED] Sent: Monday, January 02, 2006 2:29 PM To: Usman Tahir Cc: Asterisk Users List Subject: Re: [Asterisk-Users] snom Firmware 5.0. Thanks for the new firmware, finally some of the features are becoming available that make the phone more usable with Asterisk. One question though, ringer tone #2 on the Snom 360 firmware has been replaced? How can I get the old ringtone back? I was using the ringtone on phones in locations like meeting rooms. The ringtone wasn't intrusive at all, yet well audible. Now when a phone rings everybody is disturbed with a loud noise. Thanks! Remco On Thu, 22 Dec 2005, Usman Tahir wrote: Hi, Snom phones firmware 5.0 is now out. Try it if you like: http://www.snom.com/wiki/index.php/Main_Page. Regards, - Usman Tahir snom technology AG www.snom.com - ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] snom Firmware 5.0.
Title: snom Firmware 5.0. Hi, Snom phones firmware 5.0 is now out. Try it if you like: http://www.snom.com/wiki/index.php/Main_Page. Regards, - Usman Tahir snom technology AG www.snom.com - ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] connection between asterisk and cisco
HI! how are you people. i am a newbie in asterisk and voip. i need your help. the scenerio is like this. 1.all local SIP users will be connected to asterisk via IP. 2.PSTN will be connected to AS5300.pstn will give us a local prefix like 333. so any one calling at 333 will go to my as5300. 3.now i want if someone calls via PSTN to a number 333 this should go to my some sip user e.g john (connect to asterisk via ip). but only to john. 4.now when john dials to any number outside 333 range , it should be dialed to the destination via AS5300(which is connected to PSTN). and destination should see that it is called by a number 333. 5.now if all this scenerio is possible, how the asterisk server and As5300 will talk to each other. what protocol can be used between them. and what physical connection i.e like ethernet or E-1 connection between as5300 and asterisk server. 6.which billing radius server you recommend, and what kind of cards will be required in a5300. thanks a lot for reading this. and thanks for reply in advance. any other suggestions are also welcome. regards __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DNS SRV
Hi, I need to run sip on non-standard port e,g 8881 and do not want user to define this port in clients like ata or softphone. what I want, when a client sends a register request at sip server, the sip server should send him the port number OR is there a way using DNS SRV can any 1 help me out ? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SS7 with Asterisk
anyone running SS7 with Asterisk ? Please help me out. I need to know the hardware used for SS7 with Digium E1 cards... Thanks, ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SS7 support ?
Is there any digium card that support E1 with SS7 and does Asterisk support SS7 ??? any 1 who has done this ? Usman ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SS7 support ?
Is there any digium card that support E1 with SS7 and does Asterisk support SS7 ??? any 1 who has done this ? Usman ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Automon filenames
Hi ! When u enable queue monitoring application from queues.conf then u have to specify a variable named MONITOR_FILENAME in extesnions.conf just before u put the incoming call into the queue. This variable will contain the path of the filename or the filename itself as with which u want to save. If u dont specify that variable then by default it will be stored in /var/spool/asterisk/monitor directory and will be named as the unique Call-ID for that particular call. Hope it helps ! Regards, Khan. On Sun, 4 Sep 2005, Anton Krall wrote: Guys. How are filenames determined for automon and queue recordings enabled on queues.conf? I see the names have some tomestamps or something but is there a way to predefine the filenames to use? Thx! ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MOH Jittery Voice
Hi All, I am having trouble with MOH. I have downloaded the latest CVS head and when I try to call from PSTN side and play MOH on a queue then the voice breaks. However if I play the same file using Playback() application and listen to it through PSTN side then there is no problem. CVan somebody tell me how can i use Playbak or background application to be used as MOH player I am waiting for any response. Khan. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] MOH Jittery Voice
Hi Wiley, thanks for ur reply ! yes ! I am using custom music files. And they are not mp3 rather they are .wav files. Even if I use mp3 files the problem remains the same. And about transferring them I use SCP to transfer files which internally uses SSH i guess. I am not sure about MOH volume. But the problem is that why does Playback () application plays file normally while MOH cant. Is there any way that I can use Playback () as MOH ??? what else could be wrong ?? I am waiting for ur response.thanks. Khan. On Wed, 1 Jun 2005, Wiley Siler wrote: Are you using custom music files? If so, how did you transfer them to the box? If you transferred via FTP, you need to be sure you set the tranfer type to Binary before sending. Tranferring using ASCII has always hosed mp3 files for me on the * box. The net result being similar to your description. Are you using the MOH definition that has normal volume? Thanks, Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Wednesday, June 01, 2005 7:52 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] MOH Jittery Voice Hi All, I am having trouble with MOH. I have downloaded the latest CVS head and when I try to call from PSTN side and play MOH on a queue then the voice breaks. However if I play the same file using Playback() application and listen to it through PSTN side then there is no problem. CVan somebody tell me how can i use Playbak or background application to be used as MOH player I am waiting for any response. Khan. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 1800 DNIS and asterisk (HOW TO?)
Hi I have applied for Qwest 1800 termination to a T1 ISDN PRI. They tell me that I will have to program a predefined DNIS number on my switch. According to them unless asterisk returns that DNIS number no call will get through. How do I program the DNIS, is it through zaptel.conf or some other way. Is it required??. As per qwest if the 8xx # is going to be routing to an ISDN TG, DNIS is required. Will appreciate any help Regards ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Putting in an Application
Hi All! I am using Asterisk Stable 1.0.6 . Now I want to add another application like app_chanspy in it. I have downloaded its source file but how can I merge this application along with my already running asterisk ? Any comments suggestions are appreciated ... Thankyou, Usman. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Barge In With Queues
Hi ! I wanted to use Barge IN with queues. ACtually what I want to do is a SIP user comes in a queue and then goes to a SIP agent. I want any application that allows me to listen to the conversation between them. I can be a supervisor extension or anything. I have used Flash Operator Panel but it works only if two asterisk SIP extensions are calling eachother. It doesnot work in the case if one of the call comes within from a queue. Any tweaking in extesnions.conf that could help me figure this out Any useful help , comments are appreciated ... thanks. Usman. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Queue Monitor Filename Problem
Hi ! I am using queues with MOnitor Application but the thing is that Iwant to save the files starting with the Answering agent name. I have tried a lot of things but nothing seems to work. If i put Monitor application on top of dialing the agent then as soon as agent picks up the recording hangs up without recording anyhting. And if I put the Monitor application on top of Queue command then I have to specify the saving filename before I know that to which agent the call is going. ANy comments , suggestions appreciated. Thanks, Usman. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] setting up fromuser
hi all,
I have got a problem with asterisk fromuser field in sip.conf. Actually
I have got two asterisk servers communicating over sip. When a user from
Asterisk Server A calls a specific extension it is redirected to another
Asterisk Server B and that Asterisk Server B forwards it to a Soft-Switch.
Now the problem being that the other Soft-Switch takes CLI from the FROM
field of SIP Invite Packet. The Asterisk Server B puts Asterisk in the
from field when redirecting the call to Soft-Switch. If I use the
fromuser field then the CLI works but it is not dynamic. However in
originating
call from Asterisk Server A the from field is correct. So my question is
that how can I make fromuser fild in sip.conf dynamic ? Means that
whatever is coming into ${CALLERIDNUM} should be assigned to fromuser in
sip.conf. Plz assist me in this problem.
khan.
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[Asterisk-Users] Turning * Hangup off in queues
Hi ! Can somebody tell me how to turn the * Hangup option utrned off in queues. I have not used any H option but still as an agent if I press * key the user gets disconnected. Somehow it is turned on by default. Can I turn this option off In my extensions.conf I have written : exten = 8000,3,Queue(supportq|t) plz help me inthis regard ... Thanks ! Usman. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Disconnection Problem
Hi ! I am facing a serious discennection problem with asterisk queues. Now I am basically running a call center where users land from allover the place through internet. I am using asterisk queues to queue them and then they are forwarded to agents. Now what happens is that after some time randomly some calls get disconnected and Astersik basically sends the SIP BYE packet to both parties. Sometimes it happens that calls may go over 20 minutes and sometimes after just 2 minutes they get disconnected. I am using free Intel G.729 codec for testing. Can somebody guide me what might be wrong ? thanks ! usman. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SER + Asterisk Attended Call Transfer
Hi All ! First I was having trouble using attended call transfer using asterisk but thatnks to you guys I was able to make it work by adding 't' in options and applying the patch. Now I am using SER along with asterisk. SER works as SIP proxy and Asterisk performs all the necessary PBX functionalities. Can anybody guide me how to make attended call transfers work in this scenario if the SIP phone doesnot support attended call transfers. I'll be waiting for any valueable feedback. Thanks, Usman. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Working Asterisk With Vonage
Hi ! I have been working on making my asterisk server work with Vonage services. I have been able to recieve calls on my asterisk machine but i couldnt call through that account to other people. Means if i call a zap channel and then dial 1 314 652 ... then i get an error like Executing Dial(Zap/3-1, SIP/dialled number@sphone.vopr.vonage.net:5061) in new stack -- Called dialled number@sphone.vopr.vonage.net:5061 -- Got SIP response 404 Not Found back from 216.115.25.198 -- SIP/sphone.vopr.vonage.net-ec6e is circuit-busy == Everyone is busy at this time -- Executing Hangup(Zap/3-1, ) in new stack == Spawn extension (local, 192512100488, 2) exited non-zero on 'Zap/3-1' -- Hungup 'Zap/3-1' whether i dial any number ... i get the same response... and always ... Can anyone guess what might be the problem ? in sip .conf my settings are : register = username:password@sphone.vopr.vonage.net:5061 [sphone.vopr.vonage.net] type = peer fromuser = username secret = password host = asterisk machine ip:5070 fromdomain=sphone.vopr.vonage.net dtmfmode=rfc2833 nat = yes canreinvite=no In extensions.conf i have done : exten = _1.,1,Dial,SIP/[EMAIL PROTECTED]:5061,tr exten = _1.,2,Hangup Please help me in this reagard. Regards , Usman. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Transfer Problem
On Fri, 8 Oct 2004, Michael Nolan wrote: Hi ! I have checked my asterisk. It contains this patch or thBis patch is already compiled into it. can you plz guide me as to how i can make use of it ? I have pressed '#' but it doesnot give me any dial tone. Are there any special changes that need to be done in extensions.conf to make it work ? plz help me in this regard. Usman. This patch works a treat for us: http://bugs.digium.com/bug_view_page.php?bug_id=0002460 Makes all # transfers attended, but the transfer button on the phones can still be used for blind transfers from our SIP phones. Cheers, Michael On Fri, 8 Oct 2004 01:56:53 -0500 (CDT), [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi Users, I am having a prblem using attended call transfer with asterisk. Actually my sip phone does not seem to support it. Can i use attended call transfer using the dial plan ... ??? means can somehow messing up with extesnions.conf I can get attended call transfer ? And yes also is there any way I can do it with AGI scripting ? Any AGI similar examples will be a lot of help. Thanks ! Usman. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
