Re: [asterisk-users] Asterisk 13.13.1 use_callids = yes Extensions ID as CallerID
Hello, I am trying to match SRC in CSV file to clid - Caller ID to user extension number for stats purposes, however, in CSV file the SRC is the company number as set in Extensions.conf exten => _7XXX,1,Set(CALLERID(number)="Company Inc" <3788001800>) exten => _7XXX,2,Dial(SIP/voip1/13781${EXTEN:1},80) exten => _7XXX,n,Congestion() exten => _7XXX,n,Hangup() how would I change it? I have look in cdr.conf and logger.conf Thanks, From: Motty Cruz [mailto:motty.c...@gmail.com] Sent: Monday, April 03, 2017 3:52 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Cc: motty.c...@gmail.com Subject: [asterisk-users] Asterisk 13.13.1 use_callids = yes Extensions ID as CallerID Hello, In Master.csv Asterisk is loggin the Company ID set in Extensions.conf, but I configured logger.conf to log the EXT ID. For instance, the SRC in the following line should be my ext. number. Does it make sense? From my extension 4007 I called 78079745, yet in the log below the first number is 2318001800 which is the main company's number set in Extensions.conf. 2318001800 78079745 phones "ITadmin" <2318001800> SIP/4007-00015c0a SIP/voip1-00015c0b Dial SIP/voip1/78079745,80 4/3/2017 15:30 4/3/2017 15:31 2 0 NO ANSWER DOCUMENTATION 1.49E+09 Logger.conf [general] dateformat=%F %T ; ; Customize the display of debug message time stamps ; this example is the ISO 8601 date format (-mm-dd HH:MM:SS) ; ; see strftime(3) Linux manual for format specifiers. Note that there is also ; a fractional second parameter which may be used in this field. Use %1q ; for tenths, %2q for hundredths, etc. ; ;dateformat=%F %T ; ISO 8601 date format ;dateformat=%F %T.%3q ; with milliseconds dateformat = %F %T.%3q ; ISO 8601 date format with milliseconds ; ; ; This makes Asterisk write callids to log messages ; (defaults to yes) use_callids = yes -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 13.13.1 use_callids = yes Extensions ID as CallerID
Hello, In Master.csv Asterisk is loggin the Company ID set in Extensions.conf, but I configured logger.conf to log the EXT ID. For instance, the SRC in the following line should be my ext. number. Does it make sense? From my extension 4007 I called 78079745, yet in the log below the first number is 2318001800 which is the main company's number set in Extensions.conf. 2318001800 78079745 phones "ITadmin" <2318001800> SIP/4007-00015c0a SIP/voip1-00015c0b Dial SIP/voip1/78079745,80 4/3/2017 15:30 4/3/2017 15:31 2 0 NO ANSWER DOCUMENTATION 1.49E+09 Logger.conf [general] dateformat=%F %T ; ; Customize the display of debug message time stamps ; this example is the ISO 8601 date format (-mm-dd HH:MM:SS) ; ; see strftime(3) Linux manual for format specifiers. Note that there is also ; a fractional second parameter which may be used in this field. Use %1q ; for tenths, %2q for hundredths, etc. ; ;dateformat=%F %T ; ISO 8601 date format ;dateformat=%F %T.%3q ; with milliseconds dateformat = %F %T.%3q ; ISO 8601 date format with milliseconds ; ; ; This makes Asterisk write callids to log messages ; (defaults to yes) use_callids = yes -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk 13.13.1 Everyone is busy-congested at this time (1:1/0/0)
Hi, my server is running a fresh install of Asterisk 13.13.1 on CentOS 7. My extensions.conf file was mostly copied from server running Asterisk 1.8. That being said! If I dial a number and get a busy signal I get the following error: -- SIP/voipeer-084b redirecting info has changed, passing it to SIP/1007-084a -- SIP/voipeer-084b is busy == Everyone is busy/congested at this time (1:1/0/0) -- Timeout on SIP/1007-084a -- Executing [t@phones:1] Playback("SIP/1007-084a", "goodbye") in new stack > 0x7f6a62146640 -- Probation passed - setting RTP source address to 191.96.18.41:62568 -- Playing 'goodbye.slin' (language 'en') > 0x7f6a62146640 -- Probation passed - setting RTP source address to 191.96.18.41:62568 -- Executing [t@phones:2] Hangup("SIP/1007-084a", "") in new stack Sip.conf [1007] type=friend context=sip-phone call-limit=2 trustrpid=no callerid="dev1" <1007> disallow=all allow=ulaw allow=alaw username=1007 secret=X dtmfmode=rfc2833 host=dynamic mailbox=1007@default nat=force_rport,comedia Is it a codec issue? Or missed configuration? Asterisk does not know how to translate busy signal. Your help is appreciated! Thanks, -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 13.13.1
SIP packet loss is one thing, RTP packet loss is another one. One does not necessarily imply the other though, of course, both may happen for a common reason. What about audio codecs ? Is it possible to configure things so that you only have a single codec enabled all over your system (trunks, phones, ...) ? Do you still have audio issues with a single codec ? 2017-01-30 17:55 GMT+01:00 Motty Cruz <motty.c...@gmail.com>: > Fresh installed CentOS 7.3 and Asterisk 13.13.1. Download Asterisk from > here: http://downloads.asterisk.org/pub/telephony/asterisk/ > asterisk-13-current.tar.gz > > > > I continue to see errors like this: > > [2017-01-30 08:37:17] WARNING[2332]: chan_sip.c:4061 retrans_pkt: > Retransmission timeout reached on transmission 56849706-ba96a6d9-817305d0@ > 192.168.125.173 for seqno 109 (Critical Request) -- See > https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions > > Packet timed out after 32000ms with no response > > [2017-01-30 08:37:17] WARNING[2332]: chan_sip.c:4061 retrans_pkt: > Retransmission timeout reached on transmission 6e3dd238-911e2ac3-f1260152@ > 192.168.125.152 for seqno 103 (Critical Request) -- See > https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions > > Packet timed out after 32000ms with no response > > [2017-01-30 08:37:17] WARNING[2332]: chan_sip.c:4061 retrans_pkt: > Retransmission timeout reached on transmission > ed38f9c8-295a9db-c23f5242@192.168.125.144 for seqno 103 (Critical > Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+ > Retransmissions > > Packet timed out after 32000ms with no response > > [2017-01-30 08:37:17] WARNING[2332]: chan_sip.c:4061 retrans_pkt: > Retransmission timeout reached on transmission ef497d11-a81b1c00-8bfbd3bf@ > 192.168.1.244 for seqno 103 (Critical Request) -- See > https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions > > > > Before upgrading to this new server, Asterisk version 1.8 on CentOS 5.9 > hardware on both servers were similar in CPU, Memory > > > > Any support on this matter is appreciated! > > > > Thanks, > Motty > > > > *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- > boun...@lists.digium.com] *On Behalf Of *kambiz sharifi > *Sent:* Saturday, January 28, 2017 5:13 AM > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > *Subject:* Re: [asterisk-users] Asterisk 13.13.1 > > > > > > On Wed, Jan 25, 2017 at 16:00 Olivier <oza.4...@gmail.com> wrote: > > What did you exactly upgade ? Asterisk only ? Asterisk and OS ? > How did you installed Asterisk 1.8 and 13 ? From source or from package ? > > I would be curious to see what would happen after downgrading back to 1.8. > > > > 2017-01-24 21:03 GMT+01:00 Motty Cruz <motty.c...@gmail.com>: > > Hello, I recently upgraded from Asterisk 1.8 to Asterisk 13. Now users are > starting to complaint about packets loss, conversations are choppy! > > > > > PkEI don’t even know where to start looking! Choppy conversations happened > within users. I am using sip.conf > > > > [1091] > > type=friend > > context=sip-phone > > call-limit=2 > > trustrpid=no > > callerid="dev1" <1091> > > disallow=all > > allow=ulaw > > allow=alaw > > username=1091 > > secret=X > > dtmfmode=rfc2833 > > host=dynamic > > mailbox=10091@default > > nat=force_rport,comedia > > canreinvite=no > > > > extensions.conf > > exten => 1091,hint,SIP/${EXTEN} > > exten => 1091,1,Dial(SIP/${EXTEN},15,t) > > exten => 1091,2,Voicemail(${EXTEN}@default,u) > > exten => 1091,102,Voicemail(${EXTEN}@default,b) > > exten => 1091,103,Hangup > > > > [2017-01-24 10:06:33] WARNING[2287]: chan_sip.c:4061 retrans_pkt: > > Retransmission timeout reached on transmission 7c803889-63e1b3fe-c2b5ef77@ > 192.168.0.191 for seqno 156 (Critical Request) -- See > https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions > > Packet timed out after 32000ms with no response > > > > any ideas? > > > > Thanks! > > Motty > > > -- > > > _ > > > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > > > > > Check out the new Asterisk community forum at: https://community.asterisk. > org/ > > > > > > New to Asterisk? Start here: > > > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > > > > > asterisk-users mailing list > > > To UNSUBSCR
Re: [asterisk-users] Asterisk 13.13.1
CentOS 7 uses firewalld to control TCP amd UDP access. The iptables configuration will be overwritten and dynamically changed by Firewalld so don't count on the old practice of manipulating iptables directly. I recently moved our Asterisk from an old CentOS to CentOS 7 running FreePBX 14.0.1.beta2. You can add a firewalld service yp /etc/firewalld/services like mine. [root@firewall0 services]# cat Asterisk.xml asterisk Asterisk PBX You then permit this service in your interface (zones) as a service I also added a rule to get some logging on the Asterisk ports while getting things up and running. I did this all on my exterior firewall which is also a CentOS 7 system. On the Asterisk server, I do not block anything which is not a best practice but the entire internal network is very small and I consider it to be secure. You (and I) should control the interface using Firewalld with the same service and zone specifications. On 30/01/2017 12:13 PM, Motty Cruz wrote: I thought it was a firewall issues. I disabled IP Tables & Selinux, but the problem persist! I have not made changes on our firewall since the upgrade! -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle Sent: Monday, January 30, 2017 9:05 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 13.13.1 On Jan 30, 2017, at 11:55 AM, Motty Cruz motty.c...@gmail.com wrote: Fresh installed CentOS 7.3 and Asterisk 13.13.1. Download Asterisk from here: http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-13-current.tar .gz I continue to see errors like this: [2017-01-30 08:37:17] WARNING[2332]: chan_sip.c:4061 retrans_pkt: Retransmission timeout reached on transmission 56849706-ba96a6d9-817305d0@192.168.125.173 for seqno 109 (Critical Request) -- See >>> >>> Firewall? Doug -- Ron Wheeler President Artifact Software Inc email: rwhee...@artifact-software.com skype: ronaldmwheeler phone: 866-970-2435, ext 102 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 13.13.1
On 01/30/2017 at 05:55 PM Motty Cruz wrote: > Fresh installed CentOS 7.3 and Asterisk 13.13.1. Download Asterisk from here: > http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-13-current.tar.gz > > > > > I continue to see errors like this: > > [2017-01-30 08:37:17] WARNING[2332]: chan_sip.c:4061 retrans_pkt: > Retransmission timeout reached on transmission > 56849706-ba96a6d9-817305d0@192.168.125.173 for seqno 109 (Critical Request) > -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions > > Packet timed out after 32000ms with no response > > [2017-01-30 08:37:17] WARNING[2332]: chan_sip.c:4061 retrans_pkt: > Retransmission timeout reached on transmission > 6e3dd238-911e2ac3-f1260152@192.168.125.152 for seqno 103 (Critical Request) > -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions > > Packet timed out after 32000ms with no response > > [2017-01-30 08:37:17] WARNING[2332]: chan_sip.c:4061 retrans_pkt: > Retransmission timeout reached on transmission > ed38f9c8-295a9db-c23f5242@192.168.125.144 for seqno 103 (Critical Request) -- > See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions > > Packet timed out after 32000ms with no response > > [2017-01-30 08:37:17] WARNING[2332]: chan_sip.c:4061 retrans_pkt: > Retransmission timeout reached on transmission > ef497d11-a81b1c00-8bfbd3bf@192.168.1.244 for seqno 103 (Critical Request) -- > See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions > > > > Before upgrading to this new server, Asterisk version 1.8 on CentOS 5.9 > hardware on both servers were similar in CPU, Memory > > > > Any support on this matter is appreciated! Did you setup tcpdump (behind the machine) to see, if the packets really leave the machine? Can you see any answer? Regards, Michael -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 13.13.1
I thought it was a firewall issues. I disabled IP Tables & Selinux, but the problem persist! I have not made changes on our firewall since the upgrade! -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle Sent: Monday, January 30, 2017 9:05 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 13.13.1 >>> On Jan 30, 2017, at 11:55 AM, Motty Cruz motty.c...@gmail.com wrote: >>> Fresh installed CentOS 7.3 and Asterisk 13.13.1. Download Asterisk from here: http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-13-current.tar .gz >>> I continue to see errors like this: >>> [2017-01-30 08:37:17] WARNING[2332]: chan_sip.c:4061 retrans_pkt: Retransmission timeout reached on transmission 56849706-ba96a6d9-817305d0@192.168.125.173 for seqno 109 (Critical Request) -- See >>> >>> Firewall? Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started 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 -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 13.13.1
>>> On Jan 30, 2017, at 11:55 AM, Motty Cruz motty.c...@gmail.com wrote: >>> Fresh installed CentOS 7.3 and Asterisk 13.13.1. Download Asterisk from >>> here: >>> http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-13-current.tar.gz >>> >>> I continue to see errors like this: >>> [2017-01-30 08:37:17] WARNING[2332]: chan_sip.c:4061 retrans_pkt: >>> Retransmission timeout reached on transmission >>> 56849706-ba96a6d9-817305d0@192.168.125.173 for seqno 109 (Critical Request) >>> -- See >>> >>> Firewall? Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 13.13.1
Fresh installed CentOS 7.3 and Asterisk 13.13.1. Download Asterisk from here: http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-13-current.tar.gz I continue to see errors like this: [2017-01-30 08:37:17] WARNING[2332]: chan_sip.c:4061 retrans_pkt: Retransmission timeout reached on transmission 56849706-ba96a6d9-817305d0@192.168.125.173 for seqno 109 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 32000ms with no response [2017-01-30 08:37:17] WARNING[2332]: chan_sip.c:4061 retrans_pkt: Retransmission timeout reached on transmission 6e3dd238-911e2ac3-f1260152@192.168.125.152 for seqno 103 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 32000ms with no response [2017-01-30 08:37:17] WARNING[2332]: chan_sip.c:4061 retrans_pkt: Retransmission timeout reached on transmission ed38f9c8-295a9db-c23f5242@192.168.125.144 for seqno 103 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 32000ms with no response [2017-01-30 08:37:17] WARNING[2332]: chan_sip.c:4061 retrans_pkt: Retransmission timeout reached on transmission ef497d11-a81b1c00-8bfbd3bf@192.168.1.244 for seqno 103 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Before upgrading to this new server, Asterisk version 1.8 on CentOS 5.9 hardware on both servers were similar in CPU, Memory Any support on this matter is appreciated! Thanks, Motty From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of kambiz sharifi Sent: Saturday, January 28, 2017 5:13 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 13.13.1 On Wed, Jan 25, 2017 at 16:00 Olivier <oza.4...@gmail.com> wrote: What did you exactly upgade ? Asterisk only ? Asterisk and OS ? How did you installed Asterisk 1.8 and 13 ? From source or from package ? I would be curious to see what would happen after downgrading back to 1.8. 2017-01-24 21:03 GMT+01:00 Motty Cruz <motty.c...@gmail.com>: Hello, I recently upgraded from Asterisk 1.8 to Asterisk 13. Now users are starting to complaint about packets loss, conversations are choppy! PkEI don’t even know where to start looking! Choppy conversations happened within users. I am using sip.conf [1091] type=friend context=sip-phone call-limit=2 trustrpid=no callerid="dev1" <1091> disallow=all allow=ulaw allow=alaw username=1091 secret=X dtmfmode=rfc2833 host=dynamic mailbox=10091@default nat=force_rport,comedia canreinvite=no extensions.conf exten => 1091,hint,SIP/${EXTEN} exten => 1091,1,Dial(SIP/${EXTEN},15,t) exten => 1091,2,Voicemail(${EXTEN}@default,u) exten => 1091,102,Voicemail(${EXTEN}@default,b) exten => 1091,103,Hangup [2017-01-24 10:06:33] WARNING[2287]: chan_sip.c:4061 retrans_pkt: Retransmission timeout reached on transmission 7c803889-63e1b3fe-c2b5ef77@192.168.0.191 for seqno 156 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 32000ms with no response any ideas? Thanks! Motty -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started 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 -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started 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 -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 13.13.1
On Wed, Jan 25, 2017 at 16:00 Olivierwrote: > What did you exactly upgade ? Asterisk only ? Asterisk and OS ? > How did you installed Asterisk 1.8 and 13 ? From source or from package ? > > I would be curious to see what would happen after downgrading back to 1.8. > > 2017-01-24 21:03 GMT+01:00 Motty Cruz : > > Hello, I recently upgraded from Asterisk 1.8 to Asterisk 13. Now users are > starting to complaint about packets loss, conversations are choppy! > > > > > PkEI don’t even know where to start looking! Choppy conversations happened > within users. I am using sip.conf > > > > [1091] > > type=friend > > context=sip-phone > > call-limit=2 > > trustrpid=no > > callerid="dev1" <1091> > > disallow=all > > allow=ulaw > > allow=alaw > > username=1091 > > secret=X > > dtmfmode=rfc2833 > > host=dynamic > > mailbox=10091@default > > nat=force_rport,comedia > > canreinvite=no > > > > extensions.conf > > exten => 1091,hint,SIP/${EXTEN} > > exten => 1091,1,Dial(SIP/${EXTEN},15,t) > > exten => 1091,2,Voicemail(${EXTEN}@default,u) > > exten => 1091,102,Voicemail(${EXTEN}@default,b) > > exten => 1091,103,Hangup > > > > [2017-01-24 10:06:33] WARNING[2287]: chan_sip.c:4061 retrans_pkt: > > Retransmission timeout reached on transmission > 7c803889-63e1b3fe-c2b5ef77@192.168.0.191 for seqno 156 (Critical Request) > -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions > > Packet timed out after 32000ms with no response > > > > any ideas? > > > > Thanks! > > Motty > > -- > > > _ > > > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > > > > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > > > > > New to Asterisk? Start here: > > > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > > > > > 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 -- > > > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > > > New to Asterisk? Start here: > > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > > > 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 -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 13.13.1
What did you exactly upgade ? Asterisk only ? Asterisk and OS ? How did you installed Asterisk 1.8 and 13 ? From source or from package ? I would be curious to see what would happen after downgrading back to 1.8. 2017-01-24 21:03 GMT+01:00 Motty Cruz: > Hello, I recently upgraded from Asterisk 1.8 to Asterisk 13. Now users are > starting to complaint about packets loss, conversations are choppy! > > > > > I don’t even know where to start looking! Choppy conversations happened > within users. I am using sip.conf > > > > [1091] > > type=friend > > context=sip-phone > > call-limit=2 > > trustrpid=no > > callerid="dev1" <1091> > > disallow=all > > allow=ulaw > > allow=alaw > > username=1091 > > secret=X > > dtmfmode=rfc2833 > > host=dynamic > > mailbox=10091@default > > nat=force_rport,comedia > > canreinvite=no > > > > extensions.conf > > exten => 1091,hint,SIP/${EXTEN} > > exten => 1091,1,Dial(SIP/${EXTEN},15,t) > > exten => 1091,2,Voicemail(${EXTEN}@default,u) > > exten => 1091,102,Voicemail(${EXTEN}@default,b) > > exten => 1091,103,Hangup > > > > [2017-01-24 10:06:33] WARNING[2287]: chan_sip.c:4061 retrans_pkt: > > Retransmission timeout reached on transmission 7c803889-63e1b3fe-c2b5ef77@ > 192.168.0.191 for seqno 156 (Critical Request) -- See > https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions > > Packet timed out after 32000ms with no response > > > > any ideas? > > > > Thanks! > > Motty > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: https://community.asterisk. > org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 13.13.1
Hello, I recently upgraded from Asterisk 1.8 to Asterisk 13. Now users are starting to complaint about packets loss, conversations are choppy! I don't even know where to start looking! Choppy conversations happened within users. I am using sip.conf [1091] type=friend context=sip-phone call-limit=2 trustrpid=no callerid="dev1" <1091> disallow=all allow=ulaw allow=alaw username=1091 secret=X dtmfmode=rfc2833 host=dynamic mailbox=10091@default nat=force_rport,comedia canreinvite=no extensions.conf exten => 1091,hint,SIP/${EXTEN} exten => 1091,1,Dial(SIP/${EXTEN},15,t) exten => 1091,2,Voicemail(${EXTEN}@default,u) exten => 1091,102,Voicemail(${EXTEN}@default,b) exten => 1091,103,Hangup [2017-01-24 10:06:33] WARNING[2287]: chan_sip.c:4061 retrans_pkt: Retransmission timeout reached on transmission 7c803889-63e1b3fe-c2b5ef77@192.168.0.191 for seqno 156 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 32000ms with no response any ideas? Thanks! Motty -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users