# Calls dropping around 10 minutes (but only sometimes)

**URL:** <https://community.asterisk.org/t/calls-dropping-around-10-minutes-but-only-sometimes/30339>\
**Category:** Asterisk Support\
**Created:** [January 23, 2010, 9:36pm UTC](https://community.asterisk.org/t/calls-dropping-around-10-minutes-but-only-sometimes/30339 "2010-01-23T21:36:58Z")\
**Posts on this page:** 2\
**Page:** 1

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**Author:** ![system](https://global.discourse-cdn.com/asterisk/original/2X/4/475d0c889911cfb84ac4881ad6df8b523e3f9105.png) [@system](https://community.asterisk.org/u/system)\
**Post date:** [January 23, 2010, 9:36pm UTC](https://community.asterisk.org/t/calls-dropping-around-10-minutes-but-only-sometimes/30339/1 "2010-01-23T21:36:58Z")

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I have an Asterisk box setup from an AsteriskNow install disk. The whole shebang Centos based install.

I’ve got an extension (1977) which is a Grandstream SIP phone, a extension (3000) which is an X-Lite softphone, and trunks with Callcentric. The phone is connected through NAT to the Asterisk box. Port 5060 is forwarded to the Asterisk box. I’ve got these logs below, which I only partially understand. It looks to me like the connection between the extension and asterisk is lost, prompting a hang-up request from Asterisk.

```auto
[Jan 23 14:31:53] DEBUG[13002] app_macro.c: Executed application: GotoIf
[Jan 23 14:31:53] VERBOSE[13002] logger.c: -- Executing [s@macro-dialout-trunk:19] Dial("SIP/1977-09d39d18", "SIP/ud-callcentric/xxx-xxx-xxxx|300|") in new stack
[Jan 23 14:31:53] VERBOSE[13002] logger.c: -- Called ud-callcentric/xxx-xxx-xxxx
[Jan 23 14:31:53] VERBOSE[13004] logger.c: == Begin MixMonitor Recording SIP/1977-09d39d18
[Jan 23 14:31:55] VERBOSE[13002] logger.c: -- SIP/ud-callcentric-09d41de8 is making progress passing it to SIP/1977-09d39d18
[Jan 23 14:31:55] VERBOSE[13002] logger.c: -- SIP/ud-callcentric-09d41de8 is making progress passing it to SIP/1977-09d39d18
[Jan 23 14:31:56] VERBOSE[13002] logger.c: -- SIP/ud-callcentric-09d41de8 is making progress passing it to SIP/1977-09d39d18
[Jan 23 14:31:58] VERBOSE[13002] logger.c: -- SIP/ud-callcentric-09d41de8 is making progress passing it to SIP/1977-09d39d18
[Jan 23 14:32:02] VERBOSE[13002] logger.c: -- SIP/ud-callcentric-09d41de8 is making progress passing it to SIP/1977-09d39d18
[Jan 23 14:32:08] VERBOSE[13002] logger.c: -- SIP/ud-callcentric-09d41de8 answered SIP/1977-09d39d18

[Jan 23 14:41:08] NOTICE[5270] chan_sip.c: Peer '3000' is now UNREACHABLE! Last qualify: 26
[Jan 23 14:41:41] NOTICE[5270] chan_sip.c: Peer '1977' is now UNREACHABLE! Last qualify: 28
[Jan 23 14:42:24] VERBOSE[13002] logger.c: -- Executing [h@macro-dialout-trunk:1] Macro("SIP/1977-09d39d18", "hangupcall|") in new stack
[Jan 23 14:42:24] VERBOSE[13002] logger.c: -- Executing [s@macro-hangupcall:1] GotoIf("SIP/1977-09d39d18", "1?skiprg") in new stack
[Jan 23 14:42:24] VERBOSE[13002] logger.c: -- Goto (macro-hangupcall,s,4)
[Jan 23 14:42:24] DEBUG[13002] app_macro.c: Executed application: GotoIf
[Jan 23 14:42:24] VERBOSE[13002] logger.c: -- Executing [s@macro-hangupcall:4] GotoIf("SIP/1977-09d39d18", "1?skipblkvm") in new stack
[Jan 23 14:42:24] VERBOSE[13002] logger.c: -- Goto (macro-hangupcall,s,7)
[Jan 23 14:42:24] DEBUG[13002] app_macro.c: Executed application: GotoIf
[Jan 23 14:42:24] VERBOSE[13002] logger.c: -- Executing [s@macro-hangupcall:7] GotoIf("SIP/1977-09d39d18", "1?theend") in new stack
[Jan 23 14:42:24] VERBOSE[13002] logger.c: -- Goto (macro-hangupcall,s,9)
[Jan 23 14:42:24] DEBUG[13002] app_macro.c: Executed application: GotoIf
[Jan 23 14:42:24] VERBOSE[13002] logger.c: -- Executing [s@macro-hangupcall:9] Hangup("SIP/1977-09d39d18", "") in new stack
[Jan 23 14:42:24] VERBOSE[13002] logger.c: == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/1977-09d39d18' in macro 'hangupcall'
[Jan 23 14:42:24] VERBOSE[13002] logger.c: == Spawn h extension (macro-dialout-trunk, h, 1) exited non-zero on 'SIP/1977-09d39d18'
[Jan 23 14:42:24] VERBOSE[13002] logger.c: == Spawn extension (macro-dialout-trunk, s, 19) exited non-zero on 'SIP/1977-09d39d18' in macro 'dialout-trunk'
[Jan 23 14:42:24] VERBOSE[13002] logger.c: == Spawn extension (from-internal, xxx-xxx-xxxx, 4) exited non-zero on 'SIP/1977-09d39d18'
[Jan 23 14:42:24] VERBOSE[13004] logger.c: == End MixMonitor Recording SIP/1977-09d39d18
[Jan 23 14:42:27] VERBOSE[5270] logger.c: -- Registered SIP '3000' at 10.0.1.5 port 5060
[Jan 23 14:42:27] NOTICE[5270] chan_sip.c: Peer '3000' is now Reachable. (28ms / 2000ms)
[Jan 23 14:43:00] VERBOSE[5270] logger.c: -- Registered SIP '1977' at 10.0.1.5 port 5060
[Jan 23 14:43:00] NOTICE[5270] chan_sip.c: Peer '1977' is now Reachable. (31ms / 2000ms)

```

Does this by chance have anything to do with qualify=yes in my sip.conf ?

I’m new to Asterisk, so any pointers and advice is greatly appreciated. Thanks.

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<div class="post-metadata">

**Author:** ![ianplain](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/ianplain/32/1776_2.png) [@ianplain](https://community.asterisk.org/u/ianplain)\
**Post date:** [January 23, 2010, 11:02pm UTC](https://community.asterisk.org/t/calls-dropping-around-10-minutes-but-only-sometimes/30339/2 "2010-01-23T23:02:55Z")

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Hi

Well yes it does have something todo with it but also it has more to do with a network problem thats making them unreachable

Ian.