# Avoided initial deadlock

**URL:** <https://community.asterisk.org/t/avoided-initial-deadlock/3716>\
**Category:** Asterisk Support\
**Created:** [January 17, 2006, 5:56am UTC](https://community.asterisk.org/t/avoided-initial-deadlock/3716 "2006-01-17T05:56:24Z")\
**Posts on this page:** 5\
**Page:** 1

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**Author:** ![abdulzu](https://avatars.discourse-cdn.com/v4/letter/a/d78d45/32.png) [@abdulzu](https://community.asterisk.org/u/abdulzu)\
**Post date:** [January 17, 2006, 5:56am UTC](https://community.asterisk.org/t/avoided-initial-deadlock/3716/1 "2006-01-17T05:56:24Z")

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Hi All,

Some time my SIP phone taking many minutes to register the Asterisk. and if the SIP Phone is already registered and i am trying to make call its taking long time in dialing without any success.

When this issues is happening i found in log the following warrnings:

```auto
Jan 4 17:58:35 WARNING[30665] channel.c: Avoided initial deadlock for '0x9106ef8', 10 retries!
Jan 4 17:58:40 WARNING[5478] channel.c: Avoided initial deadlock for '0x9106ef8', 10 retries!
```

Please suggenst me to avoide such kind of problem.

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<div class="post-metadata">

**Author:** ![abdulzu](https://avatars.discourse-cdn.com/v4/letter/a/d78d45/32.png) [@abdulzu](https://community.asterisk.org/u/abdulzu)\
**Post date:** [January 18, 2006, 8:07am UTC](https://community.asterisk.org/t/avoided-initial-deadlock/3716/2 "2006-01-18T08:07:08Z")

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Hello guys,

Please help me to solve this problem. When this log is start apearing my SIP phones are not able to register.

I will appriciate for your replies.

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**Author:** ![stoffell](https://avatars.discourse-cdn.com/v4/letter/s/8edcca/32.png) [@stoffell](https://community.asterisk.org/u/stoffell)\
**Post date:** [January 18, 2006, 8:29pm UTC](https://community.asterisk.org/t/avoided-initial-deadlock/3716/3 "2006-01-18T20:29:57Z")

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what version of asterisk are you running?  
what phone are you using?

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<div class="post-metadata">

**Author:** ![abdulzu](https://avatars.discourse-cdn.com/v4/letter/a/d78d45/32.png) [@abdulzu](https://community.asterisk.org/u/abdulzu)\
**Post date:** [January 19, 2006, 4:54am UTC](https://community.asterisk.org/t/avoided-initial-deadlock/3716/4 "2006-01-19T04:54:21Z")

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Hello,

I am using Asterisk SVN-branch-1.2-r7231 with MySQL Realtime. and the SIP Phone is LinkSys,SIP Express,Quintum etc…

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**Author:** ![stoffell](https://avatars.discourse-cdn.com/v4/letter/s/8edcca/32.png) [@stoffell](https://community.asterisk.org/u/stoffell)\
**Post date:** [January 19, 2006, 7:03am UTC](https://community.asterisk.org/t/avoided-initial-deadlock/3716/5 "2006-01-19T07:03:22Z")

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I suggest trying asterisk v1.2.1 or 1.2.2 to make sure you’re using a stable branch and see if it works then. If it doesn’t, do a sip show settings in the cli and post your output here.