asterisk 1.8 + google voice

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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 ?

Asterisk Users 4.3 years ago 1 Answer

cap_set_proc on several modules after installation from source

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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:

Asterisk Users 4.3 years ago 3 Answers

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

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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…

Asterisk Users 4.3 years ago 0 Answers

Light indicator managed by Asterisk

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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

Asterisk Users 4.3 years ago 7 Answers

test call generator

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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…

Asterisk Users 4.3 years ago 2 Answers

Higher CPU usage on 1.6.1 than 1.4?

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Jared, Thank you for that information! Has anyone else had an experience like this?
On 12 May 2011 20:25, Jared Geiger wrote: > Hi David,
>
> When I was testing 1.6.1 for high volume channels, I couldn't get over 1000
> channels / 40 CPS without the load average spiking up due to io wait. I
> switched back to 1.4 and I can go to 3000 channels / 75 CPS with no io wait
> and a load average in the 1s. It seemed like it was caused by…

Asterisk Users 4.3 years ago 0 Answers

Discussion of Mobile SIP, Microsoft Lync

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This should be interesting, a double header Friday at 12 Noon EDT,
session 2 at 1PM EDT. 1) Pascal Doré, Media5corp. Pascal will talk about what they've been
up to in the year since his last visit. Thanks to the Asterisk mailing
list and VoIP community, their Media5fone was able to fix its g722
implementation. I like their product a lot and used it extensively on
my old iPod Touch to make and receive phone calls on our server. SIP
does rock when it works. 2) Dave Michels, VUC pillar member talks about Lync,…

Asterisk Users 4.3 years ago 0 Answers