Podrška #14988
Zatvoren
fbde, podešenje ulaznih poziva na gigaset-e, ne zvone ekstenzije
Dodano od Saša Vranić prije više od 16 godina. Izmjenjeno prije više od 16 godina.
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Opis
faruk sada poslao mail
Sasa, sada opet ne zvone telefoni 51 i 52 kad nas neko zove s vana. zvoni samo 53. molim te pogledaj ovo sto prije. ja moram sada izaci, ali cu iskljuciti firewall.
Povezani tiketi 1 (0 otvoreno — 1 zatvoren)
Izmjenjeno od Saša Vranić prije više od 16 godina
do današnjeg momenta su u gigaset-ima bile uštekane i fixne linije, pored lan-a
a sada je samo lan i asterisk....
testirali smo danas kada zvrcne sa mobitela, i zvone sve tri ext. 51, 52, 53
sada kaze da ne zvone 51 i 52, samo 53, čudno ?!?????
to bi trebalo da se zakačim i pratim šta mi cli izbacuje... i u kojem momentu, ali to treba kada faruk bude tu.
Izmjenjeno od Saša Vranić prije više od 16 godina
čudno, ovdje u fbze sam isto primjetio budalaštine sa siemens gigaset-om, ono tipa imam definisano da je
10 => 11, 12, 13
i kada zovem 10, zvone samo recimo 11 i 13 itd.. mada su svi slobodni
Izmjenjeno od Saša Vranić prije više od 16 godina
evo isjecka iz fbze
-- Executing [10@fbze:1] Dial("SIP/32-c00023d0", "SIP/11&SIP/12&SIP/13&SIP/14|400|tT") in new stack
-- Called 11
-- Called 12
-- Called 13
[Jul 29 13:49:18] WARNING[19156]: app_dial.c:1202 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)
-- SIP/11-0097cf60 is ringing
== Spawn extension (fbze, 10, 1) exited non-zero on 'SIP/32-c00023d0'
ovaj warrning je zato što nema trenutno ext.14 spojene
i sada evo zvoni samo 11 ?!?????
u rmlh nisam imao takvih problema iako isto koriste 3 slušalice na jednoj bazi
Izmjenjeno od Saša Vranić prije više od 16 godina
m1*CLI> sip show peers Name/username Host Dyn Nat ACL Port Status fbde/fbde (Unspecified) D N 0 Unmonitored 89/skype1 192.168.66.103 5060 Unmonitored 75/75 (Unspecified) D 0 Unmonitored 72/72 (Unspecified) D 0 Unmonitored 71/71 (Unspecified) D 0 Unmonitored 70/70 (Unspecified) D 0 Unmonitored 53/53 (Unspecified) D 0 Unmonitored 52/52 (Unspecified) D 0 Unmonitored 51/51 (Unspecified) D 0 Unmonitored 50/50 192.168.66.182 D 5060 Unmonitored 45/42 (Unspecified) D 0 Unmonitored 41/41 (Unspecified) D 0 Unmonitored 40/40 (Unspecified) D 0 Unmonitored 32/32 192.168.66.181 D 5060 Unmonitored 31/31 (Unspecified) D 0 Unmonitored 30/30 (Unspecified) D 0 Unmonitored 22/22 (Unspecified) D 0 Unmonitored 21/21 (Unspecified) D 0 Unmonitored 20/20 (Unspecified) D 0 Unmonitored 14/14 (Unspecified) D 0 Unmonitored 13/13 192.168.66.180 D 5060 Unmonitored 12/12 192.168.66.180 D 5060 Unmonitored 11/11 192.168.66.180 D 5060 Unmonitored 10/10 (Unspecified) D 0 Unmonitored 24 sip peers [Monitored: 0 online, 0 offline Unmonitored: 6 online, 18 offline]
Izmjenjeno od Saša Vranić prije više od 16 godina
jao, našao sam evo ovdje info
http://www.voip-info.org/wiki/view/Siemens%20Gigaset%20C450IP
- BUGS: Missing key features in firmware version 00 / 034.00 (EEPROM version 70)
Alert_Info is not supported for distinctive ring. This would not be a big problem if there was not a side effect. Using Asterisk and ring groups for example, it is desirable to use Alert_info for external call distinctive ring. Then, when the C450IP receive the Alert_Info message it simply does not ring ! There is no simple way inside Asterisk to solve this problem, except manually editing configuration files to allow for selective Alert_Info. But it is a nightmare with FreePBX as the main files are updated at each GUI refresh...
pih...
Izmjenjeno od Saša Vranić prije više od 16 godina
evo još info-a
http://www.gigaset-blog.com/archive/2007/01/siemens_gigaset_s450_ip_hybrid.html
evo pitanja
Hi,
I have very odd problem that seems to be not exclusive to me. We have now two C455IP stations set up, with two and three handsets registered to each. Two of those three handsets are set up as ring group in asterisk, so both of them ring when a given DID number rings (via two HFC isdn cards).
What happens, that sometimes only one of those two phones ring. Logs indicate that asterisk is trying to ring two, however only one comes alive. Both seem to be registered properly. The firmware is the latest, last checked about four days ago.
Of course, the more I try to reproduce the behaviour the less successful I am. This kicks in usually after a day or two, and most of the time remains until I unplug the base again. Sometimes it fixes itself after some time.
The other base (as I mentioned we have two) works fine without any problems, same fw version.
Any remote chance that any of you gentlemen would find these symptoms familiar?
Juraj
pa neki odgovor:
We have tested the S450IP to try and find out what is causing.
1. Phones not to ring occasionally.
2. Phones not being able to dial out.
3. registration problems.
Latest firmware. Two handsets each with their own profile. Registered ok.
After a fresh boot up of base.
Ring both handsets simultaneously via both profiles. Hang up on the caller end. If you never answer a handset, you can ring 100's of times and both ring.
How to simulate problem 1.
Ring both handsets. Answer a handset. Then hang up on the caller end. The handset show a termination message for a few seconds. During this display of message you cannot call the handset it reports 481 "Call Leg/Transaction Does Not Exist" for the INVITE. If the handset user hangs the call up first then the problem does not occur!
This is repeatable, every time. Both handsets have the problem.
Do this lots of times and eventually the handsets start getting very unusable. In our tests, often one handset would refuse to ring and always return the 481 error.
Once the handsets had come into this state we found that we also couldn't dial out and got error code 705 over and over. This is problem 2.
Problem 3 only occurs when the handsets are in the final unusable state. Occasional failures for no reason at all!
A reboot of the base fixes everything.
During testing we noticed that sometimes when a handset did not ring a missed call indicator did display on the phone but the phone did not ring.
Conclusions.
The base does not hang up calls properly when the other person hangs up first. A possible memory leak as a result causes the handset to eventually lock up. In our tests we managed to make this happen by making lots of calls. This would explain why some people see the problem ocurring after a period of time dependant on the number of calls.
Sometimes when dialling a test tone followed by hanging up and then dialling our speaking clock you would hear the test tone momentarily before the clock! So something is not being cleared properly in the base.
I think a trawl through the source code will be the next step. I will be informing Siemens via our Distributor in the UK and hope they fix these problems. We might have to recall some of the handsets for customers if this problem is not resolved.
Izmjenjeno od Saša Vranić prije više od 16 godina
to su upravo te stvari koje su problematične i ovdje....
samo ne znam koliki smo mi pehovi da baš uvijek nabasamo na neke verzije telefona koje imaju ove probleme.
Izmjenjeno od Saša Vranić prije više od 16 godina
firmware se automatski download-uje sa njihovog servera, to sam primjetio kada sam uključio siemense, prijavi
"postoji nova verzija software-a, download ?"
i ako potvrdim, on na bazu to postavi i onda sam pošica po slušalicama.
Izmjenjeno od Saša Vranić prije više od 16 godina
evo sada sam resetovao gigaset ovdje u fbossu i sada kada zovem 10 dobijam
m1*CLI>
-- Executing [10@fbze:1] Dial("SIP/32-c00074a0", "SIP/11&SIP/12&SIP/13&SIP/14|400|tT") in new stack
-- Called 11
-- Called 12
-- Called 13
[Jul 29 14:49:23] WARNING[12864]: app_dial.c:1202 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)
-- SIP/11-0097d0a0 is ringing
-- SIP/12-00985200 is ringing
== Spawn extension (fbze, 10, 1) exited non-zero on 'SIP/32-c00074a0'
zvone 11 i 12, a 13 ništa ?!???
Izmjenjeno od Saša Vranić prije više od 16 godina
poslao faruku info da restartuje gigaset
Saša Vranić schrieb: > reci mi faruk, da li nakon restarta gigaset-a postoji isti problem ? > > > ----- Original Message ----- > From: "Faruk Pojskic" <faruk.pojskic@fuel-boss.de> > To: "Saša Vranić" <sasa.vranic@sigma-com.net> > Sent: Tuesday, July 29, 2008 2:09:35 PM GMT +01:00 Amsterdam / Berlin / Bern / Rome / Stockholm / Vienna > Subject: Re: telefoni-problem > > Saša Vranić schrieb: > >> to ću moći pogledati ipak kada budeš tu, jer treba da se nazove itd... da pratim šta se dešava. >> >> sutra smo opet u fbze, pa se čujemo >> >> ----- Original Message ----- >> From: "Faruk Pojskic" <faruk.pojskic@fuel-boss.de> >> To: "Saša Vranić" <sasa.vranic@sigma-com.net> >> Sent: Tuesday, July 29, 2008 1:06:04 PM GMT +01:00 Amsterdam / Berlin / Bern / Rome / Stockholm / Vienna >> Subject: telefoni-problem >> >> Sasa, >> sada opet ne zvone telefoni 51 i 52 kad nas neko zove s vana. zvoni samo >> 53. molim te pogledaj ovo sto prije. ja moram sada izaci, ali cu >> iskljuciti firewall. >> >> >> > ja sam u kancelariji, pa se javi kad mogneš. > > ne, sada je opet u redu. -- Mit freundlichen Grüßen Faruk Pojskic, Dipl.-Ing. Fuel Boss Styrumerstr. 28 46045 Oberhausen Tel: +49 (0)208 4569 801 Fax: +49 (0)208 4569 802
hm...
Izmjenjeno od Saša Vranić prije više od 16 godina
objasnio faruku otprilike o čemu se radi i da se radi na otklanjanju problema, isti problem i u ze, pa je rekao ok, čim bude šta novo postavljajte.
Izmjenjeno od Saša Vranić prije više od 16 godina
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