similar to: Connection time of Transferred Calls

I'm using the the Microsoft Real-Time Communications Client API SDK using Visual Studio 6 and . NET 2003 SE to make SIP calls. Using the examples provided I can make unregistered SIP calls fine, however I am having trouble registering with Asterisk. I have to produce an XML Profile to use when registering with a registrar. The one I use is... <provision

Dears; I do not know if any had experience in using speex or ilbc with IAX and got good results, because I am facing a problem with GSM. I am facing a noise problem when I am using GSM with IAX trunk as following: IP Phone (G711) ---> Local Asterisk Box ---> IAX Trunk using GSM codec ---> Remote Asterisk Box ---> Digium Card (FXO) to terminate the call to the destination While no

Hello, In what we believe is now the final regression discovered when upgrading XenServer from Xen 4.1 to 4.3, there is an issue with RTC emulation. Win2003 SP2 is a WAET unaware operating system, whose RTC access pattern triggers Xen''s rtc_mode_no_ack logic. The result is that the domain falls into a tight loop reading RTC RegC, whoes value is always 0xc0. I have confirmed that

Hi All; Can I do transfer for the call from zap/1 to zap/2 (both are fxs)? All what I need is to add the t argument for the Dail function? And how can I transfer to be in that senario: zap/1 dial a code to transfer for zap/2, once zap/2 answered, then he can talk with zap/1 (where the third party will not hear what they are talking), then if zap/1 hanged up the handset, the call will be between

Hi List; I heared that IAX is good for NATing issues, but I do not know if it can help me in that senario: I have two Asterisks machines in different sites and both are behind NAT (both have private IP address), I need to link these two asterisks with IAX trunk (if it help really in such senario), but I do not know if it will work without doing special routing settings on the router (like

I have two Sipura 3000s, one for our main phone line the other for our fax line. I think I need to handle each device in seperate context sections. Both contexts use the s extension and things are not working as it was before I added the second Sipura for the fax line and additional context. Is it a problem to have two contexts with s extensions? What is the proper way to handle this senario?

Calculate the offset of the physical host's Real Time Clock (RTC) from UTC and pass this to virt-v2v through the libvirt XML description of the physical machine. The libvirt XML is modified to add one of the following: (no <clock/> element) - if the RTC could not be read or there was some other time calculation error. <clock offset='utc' /> - if the RTC is the

describe MyController do integrate_views describe "A common base senario" do it "no longer integrates views" do be_careful end end end integrate_views affects an attribute in the class formed by the describe factory method. The second describe generates its own class, so integrate_views is OFF at that level. I''ve already spent far, far too much

Hi All, We are using group paging and our asterisk version: 1.4.22.1, but when ever any one page to the whole group (28 extensions), the calls which are on hold on those extensions will be dropped, is there any other way to have this feature or to go with Overhead paging. Currently this has become a serious problem, can anyone through some light on this group paging senario? Thank you very much

Hi all, We have our system hosted publicly and 4 phones are connected remotely at employee's home, and when they try to do a assisted transfer to one of the employee at the main office, the call is lost. For ex: person A calls person B, person B calls person C for assisted transfer, and as soon as person B hits transfer button again to transfer person A to C, the call is lost. But in the

Trying a different approch. Senario Raid 1 setup Bootable raid drive failed Mirror has been working for almost a month and then rebooted Now can't boot mirror drive grub not mirrored from other drive. I Fixed bootable drive. Question? Can I hook up both drives and boot fixed drive then rebuilt mirror from nonbootable drive to bootable drive? Does the raid automatically rebuilt when I

Hi. I compiled the xen binaries and the kernel with the sources. DomO starts without any problems, but when I run domU, it stops and freeze. My domU is a clone of my domO. ___________________________________________ unmounting old /dev unmounting old /proc unmounting old /sys INIT: version 2.86 booting rtc: IRQ 8 is not free. modprobe: FATAL: Error inserting rtc

Hi, I am just a little confused of mian effect in the analysis of variance (ANOVA) when you include or do not include an interaction term. Let's assume a simple case of 2-way ANOVA with 2 factors A and B, each with 2 levels. If it shows that main effect for A is significant when the interaction between A and B is NOT included, and the main effect for A is NOT significant when the interaction

I want to setup a senario in which the callers hears to some music file while the phone is ringing and as soon as the line is answered the music is stopped palying. i.e. instead of the rings the caller listens to some music. Is is possible with asterisk? Kindest Muhammad Muzzamil Luqman -------------- next part -------------- An HTML attachment was scrubbed... URL:

Senario is There are 2 asterisk servers 1FXO ports connected to Panasonic PABX on extension 100 on server 1 If someone dial 100 from extension 101, call comes in on ZAP/1 call Dial,IAX2/xxxx on asterisk server 2 and from server 2 Dial/SIP/xxxx, now problem is if SIP/xxxx want to transfer this call to extension 102 then what will be the solution ? rgrds Fregi -------------- next part

Hi All, Today I faced a problem with channels getting stuck. We use asterisk 1.4.18.1, and there were 2 extensions (channels) that got stuck. When I try to do "soft hangup <channel>", it says "Requested for soft hangup" for that channel, but if we go and check once again those channels are still stuck. Also even after asterisk restart it did'nt go, finally we had to

Hi All, I have a senario where we have multiple locations and all have the ability to call using 1NXXXXXXXXX pattern, so we have created multiple contexts so the outbound goes fine, but while transfer occurs (after picking the inbound call and transfer), it uses the first 1Nxxxxxxxxx priority patterned context, like if the 3rd location is making a transfer, but 1st location have the priority

Hi Guys! I'm middle of testing my asterisk-1.8.3.2 just make sure how much call it could handle in production so following is my senario. [sipp_client]---------------[Asterisk]----------------[sipp_server] sipp_client ./sipp -sf uac_pcap.xml -d 100000 -i 172.30.254.211 -s 2000 172.30.1.47 -l 1000 -r 250 -rp 5000 -m 1000 sipp_server ./sipp -sn uas -i 172.30.245.208 In above if i set -r

Dear All I want to use asterisk for the following Senario and Need help to find a SAMPLE extension.conf Incoming call >>>>>>>>>>>Asterisk >>>>>>>>>>>>>>GSM Termination Gw first leg second leg What I want to do is putting first call leg on

Hi, Not sure exactly how I would word the subject line so appologies in advanced. We are trying to accomplish the following scenario: Location 1: PDC: fs01.loc1.example.com IP: 10.0.0.1 Location 2: SDC: fs01.loc2.example.com IP: 10.0.1.1 Clearly when we join the SDC to the PDC there is a naming conflict. The end result would be to have clients at each site resolve the fs01 name to