[asterisk-users] Segmentation fault with 13.5.0 / PJSIP 2.4.5

2015-10-11 Thread Recursive 
Dear colleagues,

I just have experienced a segmentation fault with Asterisk 13.5.0 and PJSIP 
2.4.5. Both of them have been compiled on a standard Debian Wheezy 64 bit. I 
did not apply any patch or alter the sources of Asterisk or PJSIP in any way. 
Before compiling and installing, I removed all traces of all old Asterisk and 
PJSIP versions from my system very thoroughly.

The segmentation fault occurred when I was re-registering multiple endpoints 
one after another as fast as I could do it manually (by operating the 
respective phones) and at the same time tried to make calls from these phones. 
I have attached what was happening before the segfault in the CLI; I had 
started Asterisk with -vc.

Unfortunately, I can't reproduce the situation, so if we want to know the cause 
of the problem, I guess it's necessary that a developer analyzes the attachment 
and the respective part of the source code. I think I am not (yet ?) so deep in 
Asterisk that I could help debug the problem, but if anybody tells me what to 
do I am willing to help.

I hope that the attachment won't get blocked ...

Regards,

Recursive
*CLI> -- Removed contact 'sip:hfc1z...@192.168.xx.yy:5060;line=cvx8p4nk' 
from AOR 'hFc1zQKs' due to request
 Contact hFc1zQKs/sip:hfc1z...@192.168.xx.yy:5060;line=cvx8p4nk has been deleted
 Endpoint hFc1zQKs is now Unreachable
-- Removed contact 'sip:e5jt6...@192.168.xx.yy:5060;line=7d88y4uy' from AOR 
'e5JT6VqN' due to request
 Contact e5JT6VqN/sip:e5jt6...@192.168.xx.yy:5060;line=7d88y4uy has been deleted
 Endpoint e5JT6VqN is now Unreachable
-- Attempted to remove non-existent contact 
'sip:hfc1z...@192.168.xx.yy:5060;line=cvx8p4nk' from AOR 'hFc1zQKs' by request
-- Removed contact 'sip:zyw0l...@192.168.xx.yy:5060;line=it15kl8j' from AOR 
'Zyw0lsGa' due to request
 Contact Zyw0lsGa/sip:zyw0l...@192.168.xx.yy:5060;line=it15kl8j has been deleted
 Endpoint Zyw0lsGa is now Unreachable
-- Attempted to remove non-existent contact 
'sip:e5jt6...@192.168.xx.yy:5060;line=7d88y4uy' from AOR 'e5JT6VqN' by request
-- Added contact 'sip:zyw0l...@192.168.xx.yy:5060;line=vvsqirer' to AOR 
'Zyw0lsGa' with expiration of 3600 seconds
 Contact Zyw0lsGa/sip:zyw0l...@192.168.xx.yy:5060;line=vvsqirer has been created
 Endpoint Zyw0lsGa is now Reachable
-- Executing [0172xxx@guV4QKtp_inbound:1] 
GotoIf("PJSIP/xDHoh0oP-", "0?forbidden,201") in new stack
-- Executing [0172xxx@guV4QKtp_inbound:2] 
GotoIf("PJSIP/xDHoh0oP-", "0?forbidden,201") in new stack
-- Executing [0172xxx@guV4QKtp_inbound:3] 
GotoIf("PJSIP/xDHoh0oP-", "0?forbidden,201") in new stack
-- Executing [0172xxx@guV4QKtp_inbound:4] 
GotoIf("PJSIP/xDHoh0oP-", "0?forbidden,201") in new stack
-- Executing [0172xxx@guV4QKtp_inbound:5] 
GotoIf("PJSIP/xDHoh0oP-", "0?forbidden,201") in new stack
-- Executing [0172xxx@guV4QKtp_inbound:6] 
GotoIf("PJSIP/xDHoh0oP-", "0?forbidden,201") in new stack
-- Executing [0172xxx@guV4QKtp_inbound:7] 
GotoIf("PJSIP/xDHoh0oP-", "0?101") in new stack
-- Executing [0172xxx@guV4QKtp_inbound:8] 
GotoIf("PJSIP/xDHoh0oP-", "0?101") in new stack
-- Executing [0172xxx@guV4QKtp_inbound:9] 
GotoIf("PJSIP/xDHoh0oP-", "0?101") in new stack
-- Executing [0172xxx@guV4QKtp_inbound:10] 
GotoIf("PJSIP/xDHoh0oP-", "1?101") in new stack
-- Goto (guV4QKtp_inbound,0172xxx,101)
-- Executing [0172xxx@guV4QKtp_inbound:101] 
NoOp("PJSIP/xDHoh0oP-", "xDHoh0oP") in new stack
-- Executing [0172xxx@guV4QKtp_inbound:102] 
GotoIf("PJSIP/xDHoh0oP-", "0?111") in new stack
-- Executing [0172xxx@guV4QKtp_inbound:103] 
GotoIf("PJSIP/xDHoh0oP-", "0?116") in new stack
-- Executing [0172xxx@guV4QKtp_inbound:104] 
GotoIf("PJSIP/xDHoh0oP-", "0?121") in new stack
-- Executing [0172xxx@guV4QKtp_inbound:105] 
Goto("PJSIP/xDHoh0oP-", "forbidden,201") in new stack
-- Goto (guV4QKtp_inbound,forbidden,201)
-- Executing [forbidden@guV4QKtp_inbound:201] 
Hangup("PJSIP/xDHoh0oP-", "21") in new stack
  == Spawn extension (guV4QKtp_inbound, forbidden, 201) exited non-zero on 
'PJSIP/xDHoh0oP-'
-- Added contact 'sip:hfc1z...@192.168.xx.yy:5060;line=zzizib5p' to AOR 
'hFc1zQKs' with expiration of 600 seconds
 Contact hFc1zQKs/sip:hfc1z...@192.168.xx.yy:5060;line=zzizib5p has been created
 Endpoint hFc1zQKs is now Reachable
-- Added contact 'sip:e5jt6...@192.168.xx.yy:5060;line=ydffladj' to AOR 
'e5JT6VqN' with expiration of 600 seconds
 Contact e5JT6VqN/sip:e5jt6...@192.168.xx.yy:5060;line=ydffladj has been created
 Endpoint e5JT6VqN is now Reachable
-- Removed contact 'sip:zyw0l...@192.168.xx.yy:5060;line=vvsqirer' from AOR 
'Zyw0lsGa' due to request
 Contact Zyw0lsGa/sip:zyw0l...@192.168.xx.yy:5060;line=vvsqirer has been deleted
 Endpoint Zyw0lsGa is now Unreachable
--

Re: [asterisk-users] Segmentation fault with 13.5.0 / PJSIP 2.4.5

2015-10-11 Thread Joshua Colp 

Recursive wrote:

Dear colleagues,

I just have experienced a segmentation fault with Asterisk 13.5.0 and
PJSIP 2.4.5. Both of them have been compiled on a standard Debian
Wheezy 64 bit. I did not apply any patch or alter the sources of
Asterisk or PJSIP in any way. Before compiling and installing, I
removed all traces of all old Asterisk and PJSIP versions from my
system very thoroughly.

The segmentation fault occurred when I was re-registering multiple
endpoints one after another as fast as I could do it manually (by
operating the respective phones) and at the same time tried to make
calls from these phones. I have attached what was happening before
the segfault in the CLI; I had started Asterisk with -vc.

Unfortunately, I can't reproduce the situation, so if we want to know
the cause of the problem, I guess it's necessary that a developer
analyzes the attachment and the respective part of the source code. I
think I am not (yet ?) so deep in Asterisk that I could help debug
the problem, but if anybody tells me what to do I am willing to
help.


Information is on the wiki[1] which details how to get a backtrace from 
a core dump which may be present. Otherwise it's unlikely that the 
problem can be located without exact details for reproducing it. If you 
can get a backtrace though you can file an issue on the issue tracker[2].


[1] https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace
[2] https://issues.asterisk.org/jira

--
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

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Re: [asterisk-users] same sip username with realms and chan_sip

2015-10-11 Thread Joshua Colp 

Ludovic Gasc wrote:

Hello,

same sip username with realms is possible with Asterisk ?
I've tried to have this feature with Asterisk 13.3.2 and chan_pjsip, and
now, Asterisk crashes.


Did PJSIP crash in general (it's usually a build problem if that 
happens) or was it when you were experimenting with different realms and 
such?


--
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

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Re: [asterisk-users] Segmentation fault with 13.5.0 / PJSIP 2.4.5

2015-10-11 Thread Recursive 
> 
> Information is on the wiki[1] which details how to get a backtrace from a 
> core dump which may be present. Otherwise it's unlikely that the problem can 
> be located without exact details for reproducing it. If you can get a 
> backtrace though you can file an issue on the issue tracker[2].
> 
> [1] https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace
> [2] https://issues.asterisk.org/jira
> 

Thanks for the references. I am playing around with Asterisk several months now 
and never have experienced a segfault until yesterday. I always thought the 
software would be stable (because I have heard that even large companies use 
it) and haven't compiled with the options for a core dump, so it looks like 
that we'll never know what was happening.

But I am planning to upgrade to 13.6.0 anyway, and when doing so, I'll enable 
the core dumps like shown in [1].

Regards,

Recursive

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[asterisk-users] Semicolon use in configuration?

2015-10-11 Thread Juan van Rooyen 
Hi there,

 

Hope there is a quick answer for this.

Is there an escape character in the Asterisk parser so I can use semicolon
in asterisk configuration (specifically pjsip)?

 

The reason I ask is that Spark NZ (previously Telecom NZ) uses BroadWorks,
wants the Contact User to be:


 

chan_sip never supported this, so I'm trying to get pjsip's Contact User to
do it by specifying the User portion.

However semi-colon is treated as a comment by the Asterisk parser. Adding
quotes (") around the setting doesn't seem to help.

 

Any ideas?

 

Thanks,

Juan

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Re: [asterisk-users] Semicolon use in configuration?

2015-10-11 Thread Matthew Jordan 
On Sun, Oct 11, 2015 at 8:55 PM, Juan van Rooyen
 wrote:
> Hi there,
>
>
>
> Hope there is a quick answer for this.
>
> Is there an escape character in the Asterisk parser so I can use semicolon
> in asterisk configuration (specifically pjsip)?
>
>
>
> The reason I ask is that Spark NZ (previously Telecom NZ) uses BroadWorks,
> wants the Contact User to be:
>  01234567;tgrp=01234567;trunkcontext=telecom.co...@server.ip:5060;transport=udp>
>
>
>
> chan_sip never supported this, so I’m trying to get pjsip’s Contact User to
> do it by specifying the User portion.
>
> However semi-colon is treated as a comment by the Asterisk parser. Adding
> quotes (“) around the setting doesn’t seem to help.
>

Use a '\', i.e.,

contact=sip:01234567\;tgrp=01234567\;trunkcontext=...

-- 
Matthew Jordan
Digium, Inc. | Director of Technology
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org

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