[asterisk-users] Sip registrations question
Hello everyone I want to know if it is somehow possible for asterisk to consider new registration attempts instead of matching them with old nonce Correct auth, but based on stale nonce received from 'test sip:3247@1.1.1.1;tag=79a401979bffd0d9o0' I see messages like the one above, I understand it is because of existing expire value but would like the previous expire timer to reset and issue a new registration instead Regards, Mark -- _ -- 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] G.729 codec in pass-thru mode
1. Your softphone is not sending g729 [Jun 3 13:11:27] Capabilities: us - 0x10c (ulaw|alaw|g729), *peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing)*, combined - 0x4 (ulaw) I think free version of eyebeam doesn't come with g729, try Microsip or some other with g729 codec. If it is full version, check in the advanced sip settings and allow g729 2. canreinvite should be set to yes for using pass-thru mode check this interesting article Just FYI: Can we bypass Asterisk for RTP session?http://techyatwork.blogspot.ae/2010/10/can-asterisk-bypass-rtp-and-work-like.html 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] Asterisk SIP TCP
Tried disabling qualify and changing frequency with qualify=yes already, no luck :( On Mon, Apr 15, 2013 at 10:11 PM, Mehroz Ashraf mehroz.ashra...@gmail.comwrote: I believe qualify parameters does help in doing so. Asterisk forgets about the peer info when qualify are not acknowledged. You can also check qualifyfreq to limit the number of qualifies for particular peer. On Mon, Apr 15, 2013 at 7:37 AM, Zohair Raza engineerzuhairr...@gmail.com wrote: Hello List, Is there any setting that force asterisk to auto prune or forgot the peer information if for example x number of replies are not received It keeps sending requests to the peer, I tried to turn off qualify and originating session timers to the peer but no luck Here is the message Reliably Transmitting (no NAT) to 10.200.1.55:5076: OPTIONS sip:2271@10.200.1.55:5076;transport=tcp SIP/2.0 Via: SIP/2.0/TCP 172.20.255.50:5060;branch=z9hG4bK0714eadd Max-Forwards: 70 From: Unknown sip:Unknown@172.20.255.50;tag=as6c5371b0 To: sip:2271@10.200.1.55:5076;transport=tcp Contact: sip:Unknown@172.20.255.50:5060;transport=TCP Call-ID: 433812eb21b0bb662afac65a129bb8b6@172.20.255.50:5060 CSeq: 101 OPTIONS User-Agent: ASTPBX Date: Mon, 15 Apr 2013 15:25:09 GMT Session-Expires: 80 Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [2013-04-15 11:25:09] WARNING[5183]: chan_sip.c:3386 __sip_xmit: sip_xmit of 0x7fad6c05c660 (len 609) to 10.200.1.55:5076 returned -2: Interrupted syste Before, when this retry was exceeded or connection was refused, asterisk restarted with the log message [2013-04-15 06:54:36] ERROR[5121] tcptls.c: Unable to connect SIP socket to 10.200.1.55:5075: Connection refused [2013-04-15 06:54:44] NOTICE[5167] loader.c: 2 modules will be loaded. I will produce a back trace later today and file a bug, I am using version 1.8.14.0 Please note, I have to stick with TCP because of packet loss in the network Any suggestions? Regards, Zohair Raza -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk SIP TCP
this is my secondary email Regards Zohair On Mon, Apr 15, 2013 at 10:45 PM, Mark Henry markhenry...@gmail.com wrote: Tried disabling qualify and changing frequency with qualify=yes already, no luck :( On Mon, Apr 15, 2013 at 10:11 PM, Mehroz Ashraf mehroz.ashra...@gmail.com wrote: I believe qualify parameters does help in doing so. Asterisk forgets about the peer info when qualify are not acknowledged. You can also check qualifyfreq to limit the number of qualifies for particular peer. On Mon, Apr 15, 2013 at 7:37 AM, Zohair Raza engineerzuhairr...@gmail.com wrote: Hello List, Is there any setting that force asterisk to auto prune or forgot the peer information if for example x number of replies are not received It keeps sending requests to the peer, I tried to turn off qualify and originating session timers to the peer but no luck Here is the message Reliably Transmitting (no NAT) to 10.200.1.55:5076: OPTIONS sip:2271@10.200.1.55:5076;transport=tcp SIP/2.0 Via: SIP/2.0/TCP 172.20.255.50:5060;branch=z9hG4bK0714eadd Max-Forwards: 70 From: Unknown sip:Unknown@172.20.255.50;tag=as6c5371b0 To: sip:2271@10.200.1.55:5076;transport=tcp Contact: sip:Unknown@172.20.255.50:5060;transport=TCP Call-ID: 433812eb21b0bb662afac65a129bb8b6@172.20.255.50:5060 CSeq: 101 OPTIONS User-Agent: ASTPBX Date: Mon, 15 Apr 2013 15:25:09 GMT Session-Expires: 80 Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [2013-04-15 11:25:09] WARNING[5183]: chan_sip.c:3386 __sip_xmit: sip_xmit of 0x7fad6c05c660 (len 609) to 10.200.1.55:5076 returned -2: Interrupted syste Before, when this retry was exceeded or connection was refused, asterisk restarted with the log message [2013-04-15 06:54:36] ERROR[5121] tcptls.c: Unable to connect SIP socket to 10.200.1.55:5075: Connection refused [2013-04-15 06:54:44] NOTICE[5167] loader.c: 2 modules will be loaded. I will produce a back trace later today and file a bug, I am using version 1.8.14.0 Please note, I have to stick with TCP because of packet loss in the network Any suggestions? Regards, Zohair Raza -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7942G and SEPMAC.cnf.xml and the registration
Hi, I have this for UAE, dateTimeSetting dateTemplateD/M/YA/dateTemplate timeZoneArabian Standard Time/timeZone ntps ntp name2.2.2.2/name ntpModeUnicast/ntpMode /ntp /ntps and this for Kenya http://wwp.greenwichmeantime.com/time-zone/gmt-plus-3/ my Kenya(Saudia Arabia time zone) will work for you as it is same as Kuwait dateTimeSetting dateTemplateD/M/YA/dateTemplate timeZoneSaudi Arabia Standard Time/timeZone ntps ntp name2.2.2.2/name ntpModeUnicast/ntpMode /ntp /ntps /dateTimeSetting For other references, cisco phones also have tz time zones Like I have this for US daylight time tzdata tzolsonversion2010i/tzolsonversion tzupdatertzupdater.jar/tzupdater /tzdata dateTimeSetting dateTemplateD/M/YA/dateTemplate timeZoneEastern Standard/Daylight Time/timeZone olsonTimeZoneAmerica/New_York/olsonTimeZone ntps ntp name2.2.2.2/name ntpModeUnicast/ntpMode /ntp /ntps /dateTimeSetting I have experienced some 7940,7942 7965 hanging and couldn't found any reason but that was resolved by using those in sip tcp mode instead of udp Also, I am using this patch of asterisk which made these phones work perfectly for softkeys, call transfers, 3 way conferencing and connected caller id https://issues.asterisk.org/jira/browse/ASTERISK-13145 On Sat, Mar 23, 2013 at 7:19 AM, Vladimir Mikhelson v...@mikhelson.comwrote: Bilal, Here is the respective section from my working 7906 .conf file: dateTimeSetting dateTemplateM/D/Ya/dateTemplate timeZoneCentral Standard/Daylight Time/timeZone ntps ntp name172.29.100.11/name ntpModeUnicast/ntpMode /ntp /ntps /dateTimeSetting 172.29.100.11 is my local sntp server. Hope that helps. Thank you, Vladimir On 3/20/2013 12:13 PM, bilal ghayyad wrote: Hello; The phones are registering now. I found a SEPMAC.cnf.xml file and I used sip firmware version 8.3 and I configured nat=no at sip.conf and nat to be false in xml file. But I am facing a time problem, I am in Kuwait country and the time that is appearing at the Phones screen is delayed by 3 hours. Kuwait time is GMT+3. Anyone can help? Now I am placing the following in the xml file (but I am sure it needs to be corrected, how I do not know): dateTimeSetting dateTemplateD/M/Y/dateTemplate timeZoneKuwait/timeZone ntps ntp namepool.ntp.org/name ntpModeUnicast/ntpMode /ntp /ntps /dateTimeSetting Regards Bilal Hello; I am facing a problem to let Cisco IP Phone 7942G register on Asterisk. The firmware has been downloaded from the TFTP successfully and currently I am running this load SIP42.9-3-1SR2-1S* I feel that there is a problem in the SEPMAC.cnf.xml but really I do not know which one to be used exactly. Basically, there is some effect that appears on the Phone (for example, it is appearing the extension on the button), but the Phone is not able to register. I tried to ssh or even http or https to the phone but I can not access it. Although I configured the ssh in the SEPMAC.cnf.xml as following: sshUserIdadmin/sshUserId sshPasswordcisco/sshPassword Anyone tried to register Cisco 7942G on Asterisk? Which SEPMAC.cnf.xml was used? How I can access the Phone via ssh or http to be able to see the logs and understand what is happening? By the way: this SEPMAC.cnf.xml is existed on cisco website? Is it specialized for each Phone type (does it differs from Cisco IP Phone 7940 to 7942 to 7960)? Appreciate the help as really I am sticked at this point and not able to moveforward. Thanks in advance. Regards Bilal -- _ -- 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:
[asterisk-users] Directmedia Question
Hello List, I have some doubt about direct media settings. I have an asterisk 1.8.14 instance running on 172.20.255.50, a soft phone on IP 10.100.210.51 and a gateway at 10.100.210.254 I have set both gateway and peer to directmedia=yes but still on gateway I see RTP from asterisk's IP, have tried setting nat=yes/no and also specifying localnet values but not sure where I am doing wrong. Also directrtpsetup is set to yes A sip debug and sip show peer output is here http://pastebin.com/5PwqJ1KW Please assist Thanks -- _ -- 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] Directmedia question
Hello List, I have some doubt about direct media settings. I have an asterisk 1.8.14 instance running on 172.20.255.50, a soft phone on IP 10.100.210.51 and a gateway at 10.100.210.254 I have set both gateway and peer to directmedia=yes but still on gateway I see RTP from asterisk's IP, have tried setting nat=yes/no and also specifying localnet values but not sure where I am doing wrong. Also directrtpsetup is set to yes A sip debug and sip show peer output is here http://pastebin.com/5PwqJ1KW Please assist Thanks -- _ -- 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
