Hi list!

We’re using Asterisk in a setup where you can transfer a call via the Asterisk feature, or via the transfer button on a SIP phone. Both work.

However, in my dialplan I cannot distinguish normal calls from calls made by pressing the Transfer button on a phone.

To clarify, I would like to know the difference in the call from A to B, in these two situations:

1 (normal call)
A calls B

2 (transferred call)
C calls A
A presses “Transfer” button and calls B

Does anyone know a way how to accomplish this? Is it at all possible?

Thanks in advance for any hints,

Roel

Hi,

I encountered a strange behaviour using realtime extensions… (on Asterisk 11.2)

when I use the following static dialplan, everything works as expected..:

[from-sip]
exten => 110,1,Dial(DAHDI/g0/${EXTEN})
exten => 112,1,Dial(DAHDI/g0/${EXTEN})
exten => _XXX,1,Dial(SIP/${EXTEN})
exten => _X.,1,Dial(DAHDI/g0/${EXTEN})

will say… if a sip phone calls “110″ or “112″ the call is routed into PSTN (german emergency call)
if a sip phone calls any three digit number, the call should be routet to the corresponding SIP user and if a sip phone calls any other number the call should be routed into PSTN… thats ok and works as expected.

when I change to realtime:
[from-sip]
switch => Realtime

and put the diaplan into the database id context exten priority app appdata
“1″ “from-sip” “110″ “1″ “Dial” “DAHDI/g0/${EXTEN}”
“2″ “from-sip” “112″ “1″ “Dial” “DAHDI/g0/${EXTEN}”
“3″ “from-sip” “_XXX” “1″ “Dial” “SIP/${EXTEN}”
“4″ “from-sip” “_X.” “1″ “Dial” “DAHDI/g0/${EXTEN}”

only the emergency calls work and any other call goes to DAHDI… I cant reach any other SIP phone. Even when swapping the content of the rows 3 and 4 in the database to id context exten priority app appdata
“1″ “from-sip” “110″ “1″ “Dial” “DAHDI/g0/${EXTEN}”
“2″ “from-sip” “112″ “1″ “Dial” “DAHDI/g0/${EXTEN}”
“3″ “from-sip” “_X.” “1″ “Dial” “DAHDI/g0/${EXTEN}”
“4″ “from-sip” “_XXX” “1″ “Dial” “SIP/${EXTEN}”

makes no difference… I thought, using realtime extensions would read the dialplan from top to bottom, ordered by “id”… but it seems to be ignored somehow and the extension “_X.” catches the calls before the extensionpattern “_XXX” is reached.

I _could_ avoid this be prefixing “external” numbers with a leading 0
for example… but I dont want to… as I said.. using static extension via extensions.conf the dialplan works as expected…

Am I missing something?

regards, yves

Hi, I have to create scenario like following, I have 2 sip soft phone.I configured Asterisk server on local network, on Linux.With two soft-phone , local asterisk sever, i able to communicate.Now i have communicate with other network SIP client.For that i have opened account at @sip2sip.info, they provided me credentials.Then i registered one SIP phone to local Asterisk sever and another to Sip2sip.info , Can i able to communicate with this scenario ? How i should configure extention.conf in local asterisk sever to communicate with soft-phone which registered at Sip2sip.info ??Or if you have any other idea to crate such scenario please let me know ??Please also recommend me any good “SIP Developer” group ??

Best Regards,Sakharam Thorat.

We recently set up a SIP trunk between an office in NY running Asterisk and an office in Paris (running Alcatel). All works fine if a SIP phone on the NY system talks to the Paris PBX. But if something on DAHDI (a PRI or MeetMe) talks to the Paris PBX, there’s a low-volume crackling. This isn’t clipping because it also occurs when there’s no legitimate sound. It’s sort of a mild version of what you used to get when a POTS pair had a ground short. This occurs no matter what size originates the call.

pings show round trip times of around 100ms, ranging from around 200 to 80
ms. Packet loss is zero. The fact that SIP->SIP works fine suggests the issue isn’t related to IP issues.

I tried adding a jitter buffer, but that didn’t make a difference.

I’ve tried this sending just ULAW and G722 and allowing everything, but no difference. The SDP that comes back from Paris doesn’t list any audio codecs and is:

v=0
o

Hello,

I want to offer SIP phone user a “custom fax-to-email” feature. Here is how I would describe this feature:

- for every SIP phone,a custom email address is defined
- when a SIP phone answers an incoming call (from a trunk or another SIP
endpoint), it detects the call is coming from a fax machine and then :
+ it plays a pre-recorded audio file to the receiving user (“You are now receiving a fax call, please check you email box”)
+ while at the same time, the incoming channel is forwarded to an appropriate statement within Asterisk dialplan.
- when an unanswered call is forwarded to a voicemail, the fax call is also detected and teated appropriately.


1. It is possible to play a pre-recorded audio file to the receiving user ?
If positive, how can it be done ?

2. What is the exact purpose of sip.conf faxdetect setting in this case given the assumption “faxdetect is set to yes in general section of sip.conf”. I would say the following applies:
“faxdetect for the incoming channel has no influence at all. If faxdetect is set to yes or unset in the outgoing channel, then Asterisk will jump to fax extension. If faxdetect is set to no in the outgoing channel, then Asterisk will jump to fax extension.”

Do you agree ?

3. Using CLI, is there a way to read the faxdetect parameter value of a given SIP peer ?
To me “sip show peer ” doesn’t (seem to) display this.

4. When I type “fax show settings” in, I’ve got (on an Asterisk 10 box):
CLI> fax show settings FAX For Asterisk Settings:
ECM: Enabled
Status Events: Off
Minimum Bit Rate: 2400
Maximum Bit Rate: 14400
Modem Modulations Allowed: V17,V27,V29


FAX Technology Modules:

Spandsp (Spandsp FAX Driver) Settings:
CLI>

This last line troubles me a little (did I forget to configure spandsp ?). Should I care ?


Regards