[asterisk-users] Connection dropped after 15 minutes with Deutsche Telekom
Hi list! I already had the problem last year, then it would be solved (surely from some technician by Deutsche Telekom on their servers), and now I have the problem again (but I didn't changed my Asterisk configuration). The problem: after 15 minutes will the call dropped, but only if the call is to another nation! If I just call another phone in Germany, I can speak longer than 15 minutes... Someone here is customer by Deutsche Telekom, too and solved this problem? Unfortunately, calling the customer support of Telekom does not help, then they just don't know about what they speak... As soon as I say, that I don't have a Router managing the phones, they don't know what they can do (but they say, that they have problem connecting to my Router and I have to check the DSL connection... :( ). Thanks a lot! Luca Bertoncello (lucab...@lucabert.de) -- _ -- 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] no rtp after dns query (SOLVED)
thanks for confirmation dns name in /etc/hosts & dnsmgr enabled solved my problem Dne 14/12/2016 v 13:50 Joshua Colp napsal(a): On Wed, Dec 14, 2016, at 08:47 AM, marek cervenka wrote: i found this issue https://issues.asterisk.org/jira/browse/ASTERISK-26280 but its not clear if this problem can be in chan_sip/udp created channels & pjsip module is active only for wss transport This was in RTP, so it was applicable to every channel driver that uses RTP including chan_sip. -- _ -- 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] app_queue missed calls per agent - caller hangup before timeout
hi,
i'm trying get report about missed calls per agent. i'm using queue_log
and RINGNOANSWER event
but i found problem described here
---
https://www.thirdlane.com/forum/queue-log-problem
RINGNOANSWER only happens if the call TIMES OUT ringing the agent and it
returns to the queue. If your agent has a 30 second timeout and the
caller ABANDONS the call in 5 seconds it will log an ABANDON not a
RINGNOANSWER.
This is the only time ast_queue_log is executed with RINGNOANSWER. The
subsequent code of this function goes on to autopause the agent/member
if autopause is enabled. Not something that happens when callers hang up
when ringing the agents.
/*! \brief RNA == Ring No Answer. Common code that is executed when we
try a queue member and they don't answer. */
static void rna(int rnatime, struct queue_ent *qe, char *interface, char
*membername)
{
if (option_verbose > 2)
ast_verbose( VERBOSE_PREFIX_3 "Nobody picked up in %d ms\n", rnatime);
ast_queue_log(qe->parent->name, qe->chan->uniqueid, membername,
"RINGNOANSWER", "%d", rnatime);
---
any tips howto detect missed calls where caller hangup before timeout?
tnx
Marek
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[asterisk-users] Asterisk versions supporting Path header?
The bug tracker includes several issues relating to Path (RFC 3327) support. It is not clear which version actually included the patch and which versions are working. Could anybody update these issues in Jira with a brief comment about the supported versions? https://issues.asterisk.org/jira/browse/ASTERISK-16884 original patch against chan_sip / Asterisk 1.8 Status is "Fixed", but not version is recorded, which version was this merged in? https://issues.asterisk.org/jira/browse/ASTERISK-21084 chan_pjsip Path support Satus is Fixed for v12.1.0 - is that only for chan_pjsip, or is Path also supported in chan_sip in any versions up to 12.1.0? https://issues.asterisk.org/jira/browse/ASTERISK-25666 Path header ignored (looks like a regression?) reported for 13.6.0 - which is the last version where it did work? https://jira.digium.com/browse/SWP-2484 "add Path header support to chan_sip" Internal Jira link - does this issue contain any further details about the versions supported? -- _ -- 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] no rtp after dns query
On Wed, Dec 14, 2016, at 08:47 AM, marek cervenka wrote: > i found this issue https://issues.asterisk.org/jira/browse/ASTERISK-26280 > > but its not clear if this problem can be in chan_sip/udp created > channels & pjsip module is active only for wss transport This was in RTP, so it was applicable to every channel driver that uses RTP including chan_sip. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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] no rtp after dns query
i found this issue https://issues.asterisk.org/jira/browse/ASTERISK-26280 but its not clear if this problem can be in chan_sip/udp created channels & pjsip module is active only for wss transport Dne 14/12/2016 v 12:14 marek cervenka napsal(a): hi, i have strange problem with no rtp packets from asterisk after dns query. see pcap below centos6/asterisk 13.9 + chan_sip 172.23.0.3 - asterisk 172.23.5.1/2 - voip phones any ideas/hints? 1170 25.028206000 172.23.0.3 -> 172.23.5.1 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x334508F6, Seq=49318, Time=1442112256 1171 25.045556000 172.23.5.1 -> 172.23.0.3 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x643C9869, Seq=4468, Time=716240 1172 25.045629000 172.23.0.3 -> 172.23.5.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x3566361, Seq=60990, Time=716240 1173 25.048134000 172.23.5.2 -> 172.23.0.3 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x8B41FC0A, Seq=274, Time=1442112421 1174 25.048207000 172.23.0.3 -> 172.23.5.1 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x334508F6, Seq=49319, Time=1442112416 1175 25.065362000 172.23.5.1 -> 172.23.0.3 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x643C9869, Seq=4469, Time=716400 1176 25.065441000 172.23.0.3 -> 172.23.5.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x3566361, Seq=60991, Time=716400 1177 25.068138000 172.23.5.2 -> 172.23.0.3 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x8B41FC0A, Seq=275, Time=1442112581 1178 25.068214000 172.23.0.3 -> 172.23.5.1 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x334508F6, Seq=49320, Time=1442112576 1179 25.085427000 172.23.5.1 -> 172.23.0.3 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x643C9869, Seq=4470, Time=716560 1180 25.08550 172.23.0.3 -> 172.23.5.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x3566361, Seq=60992, Time=716560 1181 25.088133000 172.23.5.2 -> 172.23.0.3 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x8B41FC0A, Seq=276, Time=1442112741 1182 25.088207000 172.23.0.3 -> 172.23.5.1 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x334508F6, Seq=49321, Time=1442112736 1183 25.099395000 172.23.5.2 -> 172.23.0.3 RTCP 94 Sender Report 1184 25.099569000 172.23.0.3 -> 172.16.1.20 DNS 78 Standard query 0xd853 A pbx.somewhere.com 1185 25.099591000 172.23.0.3 -> 172.16.1.20 DNS 78 Standard query 0xeb9f pbx.somewhere.com 1186 25.100211000 172.16.1.20 -> 172.23.0.3 DNS 94 Standard query response 0xd853 A 172.23.0.3 1187 25.105456000 172.23.5.1 -> 172.23.0.3 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x643C9869, Seq=4471, Time=716720 1188 25.108153000 172.23.5.2 -> 172.23.0.3 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x8B41FC0A, Seq=277, Time=1442112901 1189 25.125115000 172.23.5.1 -> 172.23.0.3 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x643C9869, Seq=4472, Time=716880 1190 25.128169000 172.23.5.2 -> 172.23.0.3 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x8B41FC0A, Seq=278, Time=1442113061 1191 25.145232000 172.23.5.1 -> 172.23.0.3 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x643C9869, Seq=4473, Time=717040 1192 25.148169000 172.23.5.2 -> 172.23.0.3 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x8B41FC0A, Seq=279, Time=1442113221 1193 25.165214000 172.23.5.1 -> 172.23.0.3 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x643C9869, Seq=4474, Time=717200 1194 25.168169000 172.23.5.2 -> 172.23.0.3 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x8B41FC0A, Seq=280, Time=1442113381 1195 25.185212000 172.23.5.1 -> 172.23.0.3 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x643C9869, Seq=4475, Time=717360 1196 25.188194000 172.23.5.2 -> 172.23.0.3 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x8B41FC0A, Seq=281, Time=1442113541 1197 25.205216000 172.23.5.1 -> 172.23.0.3 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x643C9869, Seq=4476, Time=717520 1198 25.208164000 172.23.5.2 -> 172.23.0.3 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x8B41FC0A, Seq=282, Time=1442113701 1199 25.225149000 172.23.5.1 -> 172.23.0.3 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x643C9869, Seq=4477, Time=717680 1200 25.228177000 172.23.5.2 -> 172.23.0.3 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x8B41FC0A, Seq=283, Time=1442113861 -- _ -- 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] no rtp after dns query
hi, i have strange problem with no rtp packets from asterisk after dns query. see pcap below centos6/asterisk 13.9 + chan_sip 172.23.0.3 - asterisk 172.23.5.1/2 - voip phones any ideas/hints? 1170 25.028206000 172.23.0.3 -> 172.23.5.1 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x334508F6, Seq=49318, Time=1442112256 1171 25.045556000 172.23.5.1 -> 172.23.0.3 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x643C9869, Seq=4468, Time=716240 1172 25.045629000 172.23.0.3 -> 172.23.5.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x3566361, Seq=60990, Time=716240 1173 25.048134000 172.23.5.2 -> 172.23.0.3 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x8B41FC0A, Seq=274, Time=1442112421 1174 25.048207000 172.23.0.3 -> 172.23.5.1 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x334508F6, Seq=49319, Time=1442112416 1175 25.065362000 172.23.5.1 -> 172.23.0.3 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x643C9869, Seq=4469, Time=716400 1176 25.065441000 172.23.0.3 -> 172.23.5.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x3566361, Seq=60991, Time=716400 1177 25.068138000 172.23.5.2 -> 172.23.0.3 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x8B41FC0A, Seq=275, Time=1442112581 1178 25.068214000 172.23.0.3 -> 172.23.5.1 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x334508F6, Seq=49320, Time=1442112576 1179 25.085427000 172.23.5.1 -> 172.23.0.3 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x643C9869, Seq=4470, Time=716560 1180 25.08550 172.23.0.3 -> 172.23.5.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x3566361, Seq=60992, Time=716560 1181 25.088133000 172.23.5.2 -> 172.23.0.3 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x8B41FC0A, Seq=276, Time=1442112741 1182 25.088207000 172.23.0.3 -> 172.23.5.1 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x334508F6, Seq=49321, Time=1442112736 1183 25.099395000 172.23.5.2 -> 172.23.0.3 RTCP 94 Sender Report 1184 25.099569000 172.23.0.3 -> 172.16.1.20 DNS 78 Standard query 0xd853 A pbx.somewhere.com 1185 25.099591000 172.23.0.3 -> 172.16.1.20 DNS 78 Standard query 0xeb9f pbx.somewhere.com 1186 25.100211000 172.16.1.20 -> 172.23.0.3 DNS 94 Standard query response 0xd853 A 172.23.0.3 1187 25.105456000 172.23.5.1 -> 172.23.0.3 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x643C9869, Seq=4471, Time=716720 1188 25.108153000 172.23.5.2 -> 172.23.0.3 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x8B41FC0A, Seq=277, Time=1442112901 1189 25.125115000 172.23.5.1 -> 172.23.0.3 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x643C9869, Seq=4472, Time=716880 1190 25.128169000 172.23.5.2 -> 172.23.0.3 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x8B41FC0A, Seq=278, Time=1442113061 1191 25.145232000 172.23.5.1 -> 172.23.0.3 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x643C9869, Seq=4473, Time=717040 1192 25.148169000 172.23.5.2 -> 172.23.0.3 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x8B41FC0A, Seq=279, Time=1442113221 1193 25.165214000 172.23.5.1 -> 172.23.0.3 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x643C9869, Seq=4474, Time=717200 1194 25.168169000 172.23.5.2 -> 172.23.0.3 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x8B41FC0A, Seq=280, Time=1442113381 1195 25.185212000 172.23.5.1 -> 172.23.0.3 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x643C9869, Seq=4475, Time=717360 1196 25.188194000 172.23.5.2 -> 172.23.0.3 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x8B41FC0A, Seq=281, Time=1442113541 1197 25.205216000 172.23.5.1 -> 172.23.0.3 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x643C9869, Seq=4476, Time=717520 1198 25.208164000 172.23.5.2 -> 172.23.0.3 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x8B41FC0A, Seq=282, Time=1442113701 1199 25.225149000 172.23.5.1 -> 172.23.0.3 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x643C9869, Seq=4477, Time=717680 1200 25.228177000 172.23.5.2 -> 172.23.0.3 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x8B41FC0A, Seq=283, Time=1442113861 -- _ -- 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
