* You are viewing the archive for May 12th, 2011

asterisk 1.8 + google voice

somewhere along the way, i noticed incoming calls from google voice are
no longer working on my asterisk 1.8.3.2 system.

When the call comes in, asterisk immediately prints on the console:
== Spawn extension (google-in, s, 2) exited non-zero on
‘Gtalk/+12153930924-f947′
[May 12 22:47:18] NOTICE[13835]: chan_gtalk.c:1977 gtalk_parser: Remote
peer reported an error, trying to establish the call anyway

the calling side just hears ringing.

i have plenty of debug info, but nothing too interesting. anyone else
having this problem ? or is it time for bug report ?

cap_set_proc on several modules after installation from source

Hello Folks,

What could be producing the following warnings on console, after an
installation from source (Asterisk 1.4.41):

[May 12 21:36:54] WARNING[15344]: loader.c:434 load_dynamic_module:
Error loading module ‘res_musiconhold.so’:
/usr/lib/asterisk/modules/res_musiconhold.so: undefined symbol:
cap_set_proc

[May 12 21:36:54] WARNING[15344]: loader.c:434 load_dynamic_module:
Error loading module ‘app_festival.so’:
/usr/lib/asterisk/modules/app_festival.so: undefined symbol:
cap_set_proc

[May 12 21:36:55] WARNING[15344]: loader.c:434 load_dynamic_module:
Error loading module ‘app_ices.so’:
/usr/lib/asterisk/modules/app_ices.so: undefined symbol: cap_set_proc
[May 12 21:36:55] WARNING[15344]: loader.c:434 load_dynamic_module:
Error loading module ‘app_mp3.so’: /usr/lib/asterisk/modules/app_mp3.so:
undefined symbol: cap_set_proc
[May 12 21:36:55] WARNING[15344]: loader.c:434 load_dynamic_module:
Error loading module ‘app_nbscat.so’:
/usr/lib/asterisk/modules/app_nbscat.so: undefined symbol: cap_set_proc
[May 12 21:36:55] WARNING[15344]: loader.c:434 load_dynamic_module:
Error loading module ‘app_externalivr.so’:
/usr/lib/asterisk/modules/app_externalivr.so: undefined symbol:
cap_set_proc

[May 12 21:36:55] WARNING[15344]: loader.c:434 load_dynamic_module:
Error loading module ‘app_dahdiras.so’:
/usr/lib/asterisk/modules/app_dahdiras.so: undefined symbol: cap_set_proc

Uname -a:
Linux eslackware 2.6.33.4-smp #2 SMP Wed May 12 22:47:36 CDT 2010 i686
Intel(R) Core(TM)2 Duo CPU E4500 @ 2.20GHz GenuineIntel GNU/Linux

gcc -v:
Reading specs from /usr/lib/gcc/i486-slackware-linux/4.4.4/specs
Target: i486-slackware-linux
Configured with: ../gcc-4.4.4/configure –prefix=/usr –libdir=/usr/lib

regarding core modules

Hi all,
I would like to know what are core modules that are used for
asterisk?

can anyone help me regarding this…

with regards,
viswavardhan

Problem with PSTN calls (Asterisk as SIP client on embedded device)

Hi

I’ve spent two days trying to solve this issue but to no prevail and I’m
hoping to get some help.

I’ve configured Asterisk as a SIP client, running on OpenWRT on an embedded
device with onboard FXS and ATA. Asterisk is connecting to an external SIP
provider on the Internet who in turn provides a PSTN gateway. I’m able to
make calls to other SIP accounts registered on the same server who are
outside my LAN. However, I can not make calls to any PSTN numbers. When
trying to make PSTN calls it sounds like the person at the other end is
immediately rejecting the call although I know this is not the case.

Firstly, I’m absolutely sure that the PSTN gateway is working because I can
make outbound PSTN calls with the same SIP account using other SIP clients
(Empathy-SIP, SIPDroid) from the same LAN. However, when registering the
same SIP account using Asterisk from OpenWRT all PSTN calls fail. Inbound
calls from PSTN numbers also fail while calls from other SIP clients on the
same server work fine. Thus, I’m fairly confident the problem is with my
Asterisk configuration.

The SIP accounts shows as registered in Asterisk. I’ve attached detailed
error logs. The log files ‘messages-pstn.log’ shows the failed (PSTN) call
and ‘messages-voip.log’ shows the successful (VOIP) call. Note that I have
replaced actual phone numbers and domain names with *** for anonymity.

I suspect perhaps a codec issue, but I haven’t been able to identify the
actual problem. Any ideas that will help me towards solving this problem is
greatly appreciated.

Regards,
Helge

Light indicator managed by Asterisk

Hello,

is there some way to make Asterisk light up a certain light on an IP-phone ?

Like MWI, the message waiting indicator can light up if there is voicemail.

Could this light, or even other lights (like BLF-buttons) be used to
give a visual notification to the user ?

For example : if a certain value is set in the Mysql-DB and Asterisk
reads out this value, can Asterisk react upon it inside the dialplan to
make a light lit up ?

2nd example : if a certain extension is called, can we perform inside
the dialplan an action that makes a light lit up on a Snom or Yealink
IP-phone ?

I don’t know if all this is at all possible, but it doesn’t harm asking
I guess…

If BLF works, then maybe more things are possible in the same way. Just
thinking outside the box here.

test call generator

Hello Everyone,

I wonder if someone could share a manual about using SIPp for Asterisk’s
testing.

I’ll be gratefull

Regards,

Elder Arohuanca
Lima – Peru

On Tue, Sep 30, 2008 at 12:59 PM, zac wolfe wrote:

> Sipp looks pretty good! I don’t know how I missed this one. This would’ve
> saved me tons of time a couple months ago.
>
> I plan on using it to load test using 2 Asterisk servers, one to initiate
> the SIP calls, the other to receive. Thanks for the tip Alex.
>
> Zac Wolfe
> Safi Systems LLC
> www.safisystems.com
>
>
> On Sat, Sep 27, 2008 at 5:58 AM, Alex Balashov wrote:
>
>> What you are looking for is SIPP: http://sipp.sourceforge.net/
>>
>> It won’t intrinsically tell you anything about the data; it’s up to you
>> to appropriate the findings. But it accomplishes the generation of
>> traffic (and dummy media!) on a technical level.
>>
>> Igor Hernandez wrote:
>>
>> > Sam Tam wrote:
>> >> Hello everyone
>> >>
>> >>
>> >>
>> >> I am trying to look for a free test call generator that will get me
>> some
>> >> stats like PDD, ASR and call quality etc on each route. As well as do
>> >> test at every interval too
>> >>
>> >>
>> >> If you know something like this please enlighten me.
>> >>
>> >> Sam
>> >>
>> >>
>> >>
>> ————————————————————————
>> >>
>> >> _______________________________________________
>> >> — Bandwidth and Colocation Provided by http://www.api-digital.com
>> >>
>> >> AstriCon 2008 – September 22 – 25 Phoenix, Arizona
>> >> Register Now: http://www.astricon.net
>> >>
>> >> asterisk-users mailing list
>> >> To UNSUBSCRIBE or update options visit:
>> >> http://lists.digium.com/mailman/listinfo/asterisk-users
>> >
>> > Hey Sam,
>> >
>> > I’ve been looking for such a tool also. I can’t seem to find a tool that
>> > does those things.
>> >
>> > If nothing comes up in the next couple of weeks I’m going to code
>> > something up, I wouldn’t mind letting you and anyone else who might be
>> > interested have the source once its done.
>> >
>> > Let me know if you find anything thats already out there in the
>> > meantime, might just save me a few hours of work.
>> >
>> > Regards,
>> >
>> >
>>
>>
>> –
>> Alex Balashov
>> Evariste Systems
>> Web : http://www.evaristesys.com/
>> Tel : (+1) (678) 954-0670
>> Direct : (+1) (678) 954-0671
>> Mobile : (+1) (706) 338-8599
>>
>> _______________________________________________
>> — Bandwidth and Colocation Provided by http://www.api-digital.com
>>
>> AstriCon 2008 – September 22 – 25 Phoenix, Arizona
>> Register Now: http://www.astricon.net
>>
>> 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
>
> AstriCon 2008 – September 22 – 25 Phoenix, Arizona
> Register Now: http://www.astricon.net
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>