search for: sippy

...ed a convenient way to test our Asterisk voice apps. The obvious tool in the Open Source world is SIPp[1]. SIPp is great! Though it's a little clumsy to use sometimes, especially if you're trying to use it to drive interactive calls like an IVR. So to make our own lives easier, we created Sippy Cup. I wanted to announce it here in the hopes that it makes your lives easier as well. Sippy Cup is an Open Source (MIT license) piece of software that allows you to define an entire SIPp load test profile in a single, simple YAML format. This includes not only the test steps, but also the load...

...ot;<sip:208.65.xxx.xxx:5061> To: <sip:1778xxxxxxx at 208.65.xxx.xxx:5060> From: <sip:604xxxxxxx at 208.65.xxx.xxx:5060>;tag=ixpa27sbhn3inu5x.o Call-ID: 550D37B3 at 208.72.xxx.xxx~o CSeq: 819 INVITE Expires: 300 Content-Disposition: session Content-Type: application/sdp User-Agent: Sippy cisco-GUID: 2851810672-711266784-2763915291-559912524 h323-conf-id: 2851810672-711266784-2763915291-559912524 Content-Length: 109 v=0 o=Sippy 223452192 0 IN IP4 74.205.216.77 s=- t=0 0 m=audio 33830 RTP/AVP 0 c=IN IP4 74.205.216.777 <-------------> --- (17 headers 6 lines) --- Sending to 20...

...containing BPA, and as of Dec. 2006 the city of San Franscisco followed suit. In March 2007 a billion-dollar class action suit was commenced against Gerber, Playtex, Evenflo, Avent, and Dr. Brown's in Los Angeles superior court for harm done to babies caused by drinking out of baby bottles and sippy cups containing BPA. So, to be certain that your baby is not exposed, use glass bottles" She suggests that if you really want a plastic bottle, get ones made from a different kind of plastic. http://www.nalgene-outdoor.com/store/SearchResult.aspx?CategoryID=10 But she really recommeds storin...

...router with TomatoUSB firmware. There is a setting "SIP Helper". I disabled this "feature" on the router then everything back to normal. There is one thing still puzzle me: It seems enable or disable this "feature" doesn't effect other SIP thunks. It could be the Sippy server use two different IP. The INVITE come from 208.65.xxx.xxx, but in its packet it try to use 74.205.216.77 as contact address. Is my guess correct? Why it does this? *Jian * On 11-01-27 04:31 PM, Chad Wallace wrote: > On Thu, 27 Jan 2011 14:52:06 -0800 > Jian Gao<jian.gao at sjgeoph...

...xxx.xxx:5061> To: <sip:160428xxxxx at 208.65.xxx.xxx:5060> From: "CID NAME"<sip:604777xxxx at 208.65.xxx.xxx:5060>;tag=kvspovbxperbwmfk.o Call-ID: 12904465 at 208.xx.xx.xx~o CSeq: 493 INVITE Expires: 300 Content-Disposition: session Content-Type: application/sdp User-Agent: Sippy cisco-GUID: 4084071434-3712422367-2859401243-560159692 h323-conf-id: 4084071434-3712422367-2859401243-560159692 Content-Length: 109 v=0 o=Sippy 153068680 0 IN IP4 74.205.xxx.xxx s=- t=0 0 m=audio 34772 RTP/AVP 0 c=IN IP4 74.205.xxx.xxx <-------------> --- (17 headers 6 lines) --- Sending to...

Hello list, scrambling different pieces of info together I've come with the following : I want to have my "register =>" statements in a MySQL-database, so I've made the following table. table ast_config : id 1 cat_metric 0 var_metric 0 commented 0 filename sip.conf category general var_name register var_val username:password at sip.provider.net In ext_config

...147.183.18:5060 > Received Address: 204.147.183.18:5060 > NAT Support: RFC3581 > Audio IP: XX.XX.XX.XX (local) > Our Tag: as78cfb201 > Their Tag: da6aae9eb017f29b6c9de270fb85c352 > SIP User agent: Sippy > Original uri: sip:204.147.183.55:1024 > Caller-ID: XXXXXXXXXX > Need Destroy: 0 > Last Message: Rx: ACK > Promiscuous Redir: No > Route: > sip:204.147.183.18;ftag=da6aae9eb017f29b6c9de270fb85c352;lr=on >...

I have been running Asterisk for years on a machine with a public IP. Most recently, I have been running 1.2.17, from the day it came out, with no (noticeable) problems. Yesterday, I switched over to a new server that is on the same public subnet, one higher than the original server. I built 1.2.17 from source on that machine (as I did on the old server). My firewall on the new machine is

...I am posting my configuration here, maybe you can shed some light on me and tell me how to do it. Do I need to use trunks in my configuration? Thanks in advance ======================================extensions.conf [default] include => incoming include => outgoing include => internal [sippy] ;exten => 1131,1,Dial(SIP/sipper1) ;exten => 1131,2,Hangup ;exten => 1132,1,Dial(SIP/sipper2) ;exten => 1132,2,Hangup [incoming] ;exten => _2149XX[50-99],1,Dial(IAX2/Elvis&IAX2/Batman&IAX2/superman) exten => _2149XX[50-99],1,Goto(internal,1133) exten => _2149XX[50-99]...

...e successful. I've been comparing Asterisk debug from both systems as well as wireshark captures, but I can't figure out why Asterisk is not sending the Linksys ATA's IP address. Broadvox uses a Sonus switch and gateway with separate IP addresses for SIP and media. Affinity uses "Sippy" (?) with a common IP for SIP and media. I believe I've already covered all the possible configuration scenarios. I just can't get the right detail out of Asterisk to determine if this is an Asterisk issue, or an ITSP issue. Using bug ID#16705 as a guide, I patched this version, as w...

...rport=5060 From: <sip:2063161626 at 209.221.186.98>;tag=as11117608 To: <sip:2063161626 at 209.221.186.98>;tag=5fceb36a80dbc27aca680924f1b8b505-19ad Call-ID: 5ada9ee829ddb4d311c5cb092b8d3cf5 at 209.221.186.50 CSeq: 104 REGISTER Contact: <sip:s at 209.221.186.51>;expires=300 Server: Sippy Softswitch v2.0.80 Content-Length: 0

http://wiki.centos.org/HowTos/SoftwareRAIDonCentOS5 Comments and corrections please.

chan_sip.c6524 reload_config= unable to get ip address from asterisk, sip disabled The ip address is working fine, Internet works great. Can anyone help...Thanks