Re: [Asterisk-Users] mISDN Problem

1 hisax > slhc6240 1 isdn > mISDN_l2 36064 0 > w6692pci 23692 0 > mISDN_core 74944 2 mISDN_l2,w6692pci Mit freundlichen Grüssen Benoit Panizzon -- I m p r o W a r e A

[Asterisk-Users] misdn timeout?

Hi all I have a very strange problem here... I use a hfc-s card with mISDN in NT mode with an ISDN Phone connected. When I make a call, the phone rings two or three times and then misdn runs into a timeout... I don't know where to set that timeout, but it's way to short for the called to pick u

Re: [Asterisk-Users] misdn timeout?

> P[ 1] I IND :TIMEOUT oad:0010618115711 dad:4680041618269314 > > # Sudden Timeout?!? Uhm, I just found the problem myself after a bit more testing. Apparently my phone has a timeout of about 10 seconds in which it waits for any reply. Well you head that it's ringing, but it's not signaled as su

Re: [Asterisk-Users] bristuff for * 1.2.6/zaptel 1.2.5

me (on this list) that a new PRE bristuff was about to come out, supporting 1.2.6. For now I have switched to mISDN. Mit freundlichen Grüssen Benoit Panizzon -- I m p r o W a r e A G-System Services __ Zurlindenstrasse 29

Re: [Asterisk-Users] bristuff for * 1.2.6/zaptel 1.2.5

ab.) Ok, I miss functions like setting the time of the phones, call forwarding etc... but I suppose not all of these would also work on the ZapHFC drivers... Mit freundlichen Grüssen Benoit Panizzon -- I m p r o W a r e A G-Syst

[Asterisk-Users] zaphfc NT Mode. Extension not recognized...

Hi all I finaly set up a second * with two ZapHFC Cards. One in TE the other in NT mode. So I have a 1.2.5 Asterisk to run Meetme etc... and a 1.2.4 Asterisk to run all that Zaptel stuff. First I used mISDN on 1.2.5 which worked, but sometimes had strange behaviour. So my hope was that zaptel

[Asterisk-Users] Got SIP response 302 "Moved temporarely"

of the one the SIP provider tryes to redirect to. Any known issues? Mit freundlichen Grüssen Benoit Panizzon -- I m p r o W a r e A G-System Services __ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln

[Asterisk-Users] extensions.conf - switch => statement?

uld 'share the dialplan between these machines' as 'switch' is explained in the examples. But how do I do that exactly? Mit freundlichen Grüssen Benoit Panizzon -- I m p r o W a r e A G-System Services __ Zurlindenstra

Re: [Asterisk-Users] zaphfc NT Mode. Extension not recognized...

Hi all I managed to figure out where the problem is... zapata.conf channel => has to bee the last statement per channel definition. So if you specify overlapdial=yes after channel => this has no effect. You need to stop and restart asterisk after changes to zapata.conf. Reload does not seam to

[Asterisk-Users] Dial Plan Problem with extensions ringing multiple phones connected on different * servers

tly use that statement. Mit freundlichen Grüssen Benoit Panizzon -- I m p r o W a r e A G-System Services __ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 PrattelnFax +41 61 826 93 01 Schweiz

Re: [Asterisk-Users] How to set busy

On Sunday 09 April 2006 06:02, Miles Scruggs wrote: > For multiline phones how do you set SIP channels to busy. For instance > if SIP/101 is on a call then dial would return busy. Right now it just > starts ringing on line X, and stacks up from there. I suppose incominglinit=1 in the sip.conf of

[asterisk-users] chan_sip.c:3641 retrans_pkt: Retransmission timeout

e not accepted by asterisk 1.8.10.0 ? The other way round (asterisk => c3) the calls work fine. Regards Benoit Panizzon -- I m p r o W a r e A G- __ Zurlindenstrasse 29 Tel +41 61 826 93 07 CH-4133 Pr

[asterisk-users] Invite + decreasing sequence number => 500 Error?

e sumbled over an asterisk bug. Is there anyone who knows? Benoit Panizzon -- I m p r o W a r e A G- __ Zurlindenstrasse 29 Tel +41 61 826 93 07 CH-4133 PrattelnFax +41 61 826 93 02 Schweiz

Re: [asterisk-users] Invite + decreasing sequence number => 500 Error?

is wrong they say. Well I'll quote them the _MUST_ part of section 22.2 Thanks Benoit Panizzon -- I m p r o W a r e A G- __ Zurlindenstrasse 29 Tel +41 61 826 93 07 CH-4133 Pratteln

Re: [asterisk-users] Invite + decreasing sequence number => 500 Error?

ed Call-ID: A220DA56@7f33ff47 CSeq: 24994731 INVITE (Asterisk) SIP/2.0 500 Server error Well as I see it, the C3 PBX just generates plain random CSeq Numbers. Regards Benoit Panizzon -- I m p r o W a r e A G- __ Zurlindenstrasse 29

[asterisk-users] How to log caller IP address in the CDR?

ell for this case it is too late now. But is there a way to get the IP Address of the SIP Client being logged in each CDR? Kind regards Benoit Panizzon -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- N

Re: [asterisk-users] How to log caller IP address in the CDR?

sure I now have set: alwaysauthreject=yes And got a script to scan the logfile all 15min to firewall IP addresses which excessively try to login. You're always smarter after the incident :-/ Benoit Panizzon -- _ -- Bandwidt

[asterisk-users] cseq decreasing => 500 Server Error

his problem or able to explain this observation? Mit freundlichen Grüssen Benoit Panizzon -- I m p r o W a r e A G- __ Zurlindenstrasse 29 Tel +41 61 826 93 07 CH-4133 PrattelnFax +41 61 826 93 02 S

[asterisk-users] Prevent Asterisk from setting CALLERID(name) or unsetting CALLERID(name)

LLERID(name)=""} but that resulted in From: "" and the displaying of this empty string on the subscribers phone. Is there a way to completely remove the CALLERID(name) like (UNSET({CALLERID(name))? Kind regards Ben

[asterisk-users] Early audio SIP sequence order question

he carrier have signaled 183 Session Progress instead of 180 Ringing? Kind regards Benoit Panizzon -- I m p r o W a r e A G- __ Zurlindenstrasse 29 Tel +41 61 826 93 07 CH-4133 PrattelnFax +41 61 82

Re: [asterisk-users] Early audio SIP sequence order question

k pass audio of it didn't yet receive a 183 or 200 message, or is the carrier doing wrong in sending early audio without 183? Kind regards Benoit Panizzon -- I m p r o W a r e A G- __ Zurlindenstrasse 29 Tel +

Re: [asterisk-users] Early audio SIP sequence order question

f you dial more than one endpoint and more than one is sending early audio, which one do you forward? I think nobody tought about that issue. Well as long as one is being forwarded that would be ok for our case :-) Kind regards Benoit Panizzon -- I m p r o W a

[asterisk-users] SIP Diversion RDNIS - how to get reason parameter?

Hi out there To play the correct announcement in app_voicemail I whould be able to read the SIP Diversion Reason which ist sent by another PBX: Invite contains: Diversion: ;reason=no- answer;privacy=off;counter=1 Asterisk Logs: RDNIS for this call is is +41315995003 (reason no-answer;privacy=

Re: [asterisk-users] SIP Diversion RDNIS - how to get reason parameter?

just put in a temporary variable __SIPDIVERSIONREASON but not in a variable useable in the dialplan. Kind regards Benoit Panizzon -- I m p r o W a r e A G- __ Zurlindenstrasse 29 Tel +41 61 826 93 07 CH-4133

[asterisk-users] Asterisk Voicemail Realtime and 'VirtualBoxing'

time. Is there a way to have the email settings per voicemail context together with a realtime vm config? Mit freundlichen Grüssen Benoit Panizzon -- I m p r o W a r e A G- __ Zurlindenstrasse 29 Tel +41

[asterisk-users] VoiceMail customizing

how messages are played via voicemail.conf? Mit freundlichen Grüssen Benoit Panizzon -- I m p r o W a r e A G- __ Zurlindenstrasse 29 Tel +41 61 826 93 07 CH-4133 PrattelnFax +41 61 826 93 02 Schweiz

[asterisk-users] app_voicemail.c how to enable debugging?

ep.gsm In case there is an unavailable message. Where do I have to poke at the source? Kind regards Benoit Panizzon -- I m p r o W a r e A G- __ Zurlindenstrasse 29 Tel +41 61 826 93 07 CH-4133 PrattelnFax

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