somewhere along the way, i noticed incoming calls from google voice are no longer working on my asterisk 220.127.116.11 system. When the call comes in, asterisk immediately prints on the console: == Spawn extension (google-in, s, 2) exited non-zero on Gtalk/+12153930924-f..
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: loader.c:434 load_dynamic_module: Error loading module res_musiconhold.so: /usr/lib/asterisk/modules/res_musiconhold…
I would like to know what are core modules that are used for
can anyone help me regarding this…
Hi Ive spent two days trying to solve this issue but to no prevail and Im hoping to get some help. Ive 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 provi..
Everyone, I wonder if someone could share a manual about using SIPp for Asterisks testing. Ill be gratefull Regards, Elder Arohuanca Lima – Peru On Tue, Sep 30, 2008 at 12:59 PM, zac wolfewrote: > Sipp looks pretty good! I dont know how I missed t..
I am looking ConfBridge for 1.8 version of asterisk. How could i obtain and install with 1.8..
Jared, Thank you for that information! Has anyone else had an experience like this? On 12 May 2011 20:25, Jared Geigerwrote: > David, > > When I was testing 1.6.1 for high volume channels, I couldnt get over 1000 > channels/ 40 CPS without the load aver..
Is there any way by which we can put multiple calls into hold with asterisk.
like A to B.
then C to B and A on hold.
then D to B now C ,A on hold like..
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 theyve been up to in the year since his last visit. Thanks to the Asterisk mailing list and VoIP communi..