[asterisk-users] dahdi configuration issue
Hello List, I have configure 2 sangoma card each with 8 PRI lines with dahdi 2.6 the problem is i can see all channels configured in dahdi_cfg 480 channels configured but when I see /dev/dahdi i can only see 240 channels. what could be problem I am using it wanrouter and when I put PRI in new card i only got calls on new line that means one of the card is inactive at same time all the lines and alarms are okay only suspected thing is /dev/dahdi. is there nany setting in linux or kernel level which need to be set for solve this issue. any help appreciated. Thanking You --Dhaval -- _ -- 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] dahdi configuration issue
Did you open a ticket at Sangoma-Site? What wanpipe driver version do you use? Is it a production machine? Or can you test it in that way, that you crossover lines from one card to the other? Am 04.09.2013 10:48, schrieb DHAVAL INDRODIYA: Hello List, I have configure 2 sangoma card each with 8 PRI lines with dahdi 2.6 the problem is i can see all channels configured in dahdi_cfg 480 channels configured but when I see /dev/dahdi i can only see 240 channels. what could be problem I am using it wanrouter and when I put PRI in new card i only got calls on new line that means one of the card is inactive at same time all the lines and alarms are okay only suspected thing is /dev/dahdi. is there nany setting in linux or kernel level which need to be set for solve this issue. any help appreciated. Thanking You --Dhaval -- _ -- 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 crash
On Tue, Sep 3, 2013 at 4:17 AM, Deka, Rajib IN MAA SL rajib.d...@siemens.com wrote: In our lab asterisk has crashed due to some unknown reason and it has been restarted by safe_asterisk service. But before crash we can see lots of below log entry (asterisk version 1.8.9.3). That is quite old. Lots of bugs (and several security issues) have been fixed since then. Try the latest in the 1.8 branch. For the crash , follow the instructions here https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace and gather a backtrace after recompiling with the required options. (preferably after upgrading to the latest 1.8, as there may have been improvmen Sep 3 07:55:21] WARNING[16287] res_rtp_asterisk.c: RTP Transmission error of packet to [2002:c117:a683::c117:a683]:20940: Address family not supported by protocol chan_sip.c: Purely numeric hostname, and not a peer--rejecting! These messages alone don't show the whole picture. https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information Collect a log with VERBOSE and DEBUG turned up to level 5, SIP debug turned on, and pastebin that. I'd wait until after you test with the latest in 1.8 -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 Check us out at: http://digium.com http://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
[asterisk-users] Macedonian DID
Hi, I searched a lot last few days but I am uanble to find a DID number in Macedoania. However no luck. any ideas about a provider ? Thanks, Peter -- _ -- 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] OpenVox G400P network registration problems
Is anybody intimately familiar with the OpenVox G400P card, or the Quectel M20 RF modules fitted to it? I am having a strange network connectivity issue with just such a card, as follows: The card was previously used with four O2 SIMs, and -- once I mastered creating message PDUs! -- worked beautifully, save for the fact that O2's definition of unlimited as in text messages turned out not to be the same as that found in the Oxford English Dictionary :( Replacement SIM cards were duly ordered, and this is when the problem has manifested itself. Span 1 will not register a T-Mobile SIM. Issuing AT+COPS=? shows only O2 and Vodafone available as operators on this span. Issuing the same command on any other span shows Orange, T-Mobile, O2 and Vodafone available. The SIM however worked properly in a mobile phone handset. Performing gsm power off 1, gsm power off 2, swapping the SIMs between these spans and then performing gsm power on 1 and gsm power on 2 results in the recalcitrant SIM registering on span 2, and the SIM formerly from span 2 not registering on span 1. I'm guessing the Quectel M20 GSM module on span 1 has got itself into a strange state; because it was also necessary to issue gsm show span 1 to read the result of the last AT command (on other spans, the result appears in the Asterisk CLI). Do you know of a way of hard-resetting it? (The obvious ATZ does not work, neither does gsm power off 1 followed by gsm power on 1). Software versions: Debian GNU/Linux 6.0.6 Asterisk 1.8.11-cert5 Dahdi 2.6.1+2.6.1 Chan_extra 2.0.5 (Yes, these are all a bit out-of-date; but they worked before. All I did was swap over the SIM cards.) -- 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] Macedonian DID
Am 04.09.2013 15:36, schrieb Zyumbilev, Peter: I searched a lot last few days but I am uanble to find a DID number in Macedoania. However no luck. any ideas about a provider ? didlogic.com had some a couple months ago, but they only lasted for a few weeks, probably offered by an individual and not a telco, then they were taken offline by the telco/regulator. I guess you'll have to wait a few years while this country progresses. But you can always get a premium number there (pay per minute/call) if that helps. BTW, asterisk-biz might be a better list for such requests. -- _ -- 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] Macedonian DID
On 04/09/2013 19:31, Markus wrote: few years while this country progresses. But you can always get a premium number there (pay per minute/call) if that helps. Thanks :-) Do you know any good premium provider there ? Peter -- _ -- 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] 回复: Fw： OpenVox G400P network registration problems
Hi, This is tech-support from OpenVox, would you mind to send email to tim.j...@openvox.cn for more details about G400P issue? Or contact me via IM below for better communication. Regards, MSN: tim.j...@msn.cn Gtalk: tim.june...@gmail.com Skype: tim.jjune OpenVox Communication Co. Ltd. Quick Support: http://wiki.openvox.cn/index.php/OpenVox_Quick_Support -- Original -- From: A J Stilesasterisk_l...@earthshod.co.uk Date: Wed, Sep 4, 2013 11:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com; Subject: [asterisk-users] OpenVox G400P network registration problems Is anybody intimately familiar with the OpenVox G400P card, or the Quectel M20 RF modules fitted to it? I am having a strange network connectivity issue with just such a card, as follows: The card was previously used with four O2 SIMs, and -- once I mastered creating message PDUs! -- worked beautifully, save for the fact that O2's definition of unlimited as in text messages turned out not to be the same as that found in the Oxford English Dictionary :( Replacement SIM cards were duly ordered, and this is when the problem has manifested itself. Span 1 will not register a T-Mobile SIM. Issuing AT+COPS=? shows only O2 and Vodafone available as operators on this span. Issuing the same command on any other span shows Orange, T-Mobile, O2 and Vodafone available. The SIM however worked properly in a mobile phone handset. Performing gsm power off 1, gsm power off 2, swapping the SIMs between these spans and then performing gsm power on 1 and gsm power on 2 results in the recalcitrant SIM registering on span 2, and the SIM formerly from span 2 not registering on span 1. I'm guessing the Quectel M20 GSM module on span 1 has got itself into a strange state; because it was also necessary to issue gsm show span 1 to read the result of the last AT command (on other spans, the result appears in the Asterisk CLI). Do you know of a way of hard-resetting it? (The obvious ATZ does not work, neither does gsm power off 1 followed by gsm power on 1). Software versions: Debian GNU/Linux 6.0.6 Asterisk 1.8.11-cert5 Dahdi 2.6.1+2.6.1 Chan_extra 2.0.5 (Yes, these are all a bit out-of-date; but they worked before. All I did was swap over the SIM cards.) -- 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-users3615_sign(11-03-11-41(11-04-14-28-46).jpg-- _ -- 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 crash
Yes we can reproduce this crash scenario by running calls between portsip and Xlite soft phones. The issue we have observed is CODEC translation between iLBC and alaw with following warning messages, [Sep 2 15:59:53] WARNING[24418] channel.c: Codec mismatch on channel SIP/18202-0002d011 setting write format to ilbc from ulaw native formats 0x4 (ulaw) [Sep 2 15:59:53] WARNING[24418] channel.c: Unable to find a codec translation path from 0x4 (ulaw) to 0x400 (ilbc) [Sep 2 15:59:53] WARNING[24418] chan_sip.c: Asked to transmit frame type ilbc, while native formats is 0x4 (ulaw) read/write = 0x4 (ulaw)/0x4 (ulaw) [Sep 2 15:59:53] WARNING[24418] channel.c: Codec mismatch on channel SIP/18202-0002d011 setting write format to ilbc from ulaw native formats 0x4 (ulaw) [Sep 2 15:59:53] WARNING[24418] channel.c: Unable to find a codec translation path from 0x4 (ulaw) to 0x400 (ilbc) [Sep 2 15:59:53] WARNING[24418] chan_sip.c: Asked to transmit frame type ilbc, while native formats is 0x4 (ulaw) read/write = 0x4 (ulaw)/0x4 (ulaw) [Sep 2 15:59:53] WARNING[24418] channel.c: Codec mismatch on channel SIP/18202-0002d011 setting write format to ilbc from ulaw native formats 0x4 (ulaw) [Sep 2 15:59:53] WARNING[24418] channel.c: Unable to find a codec translation path from 0x4 (ulaw) to 0x400 (ilbc) [Sep 2 15:59:53] WARNING[24418] chan_sip.c: Asked to transmit frame type ilbc, while native formats is 0x4 (ulaw) read/write = 0x4 (ulaw)/0x4 (ulaw) [Sep 2 15:59:53] WARNING[24418] channel.c: No path to translate from SIP/18252-0002d010 to SIP/18203-0002d01e [Sep 2 15:59:53] WARNING[24418] channel.c: Can't make SIP/18252-0002d010 and SIP/18203-0002d01e compatible [Sep 2 15:59:53] WARNING[24418] features.c: Bridge failed on channels SIP/18252-0002d010 and SIP/18203-0002d01e We can reproduce the problem as below, 1. Call between Xlite(iLBC) to portsip(G711), RTP through asterisk. 2. portsip attended transfer the call to another portsip client 3. on complete transfer asterisk crashes (then started by safe_asterisk) with above warning. FYI, we have not installed asterisk with iLBC support. We will try to upgrade asterisk and try to reproduce this scenario. Regards Rajib -- Message: 13 Date: Wed, 4 Sep 2013 09:28:12 -0500 From: Rusty Newton rnew...@digium.com Subject: Re: [asterisk-users] Asterisk crash To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: cagwdfcvp_vtcpwgw_0zspcqxiflqsrhthnl4c0zfuf92dud...@mail.gmail.com Content-Type: text/plain; charset=UTF-8 On Tue, Sep 3, 2013 at 4:17 AM, Deka, Rajib IN MAA SL rajib.d...@siemens.com wrote: In our lab asterisk has crashed due to some unknown reason and it has been restarted by safe_asterisk service. But before crash we can see lots of below log entry (asterisk version 1.8.9.3). That is quite old. Lots of bugs (and several security issues) have been fixed since then. Try the latest in the 1.8 branch. For the crash , follow the instructions here https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace and gather a backtrace after recompiling with the required options. (preferably after upgrading to the latest 1.8, as there may have been improvmen Sep 3 07:55:21] WARNING[16287] res_rtp_asterisk.c: RTP Transmission error of packet to [2002:c117:a683::c117:a683]:20940: Address family not supported by protocol chan_sip.c: Purely numeric hostname, and not a peer--rejecting! These messages alone don't show the whole picture. https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information Collect a log with VERBOSE and DEBUG turned up to level 5, SIP debug turned on, and pastebin that. I'd wait until after you test with the latest in 1.8 -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 Check us out at: http://digium.com http://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
