Re: [Asterisk-Users] Swissvoice IP10S centralized phonebook
This is the information I got from Swissvoice support, I didn't tried yet, but if it can helps. How to use an external phone book IP10S phone supports access to Cisco Phone Book but not all functionalities. The IP10 uses his own interface to access to the Phone Book. If you want to connect to your remote phone book, you have to do the following actions: First, copy the URL under Search by name in a Web browser, for example: http://192.168.1.5/cisco/directory/searchDirectory.php You are going to have a XML file display in the Web Browser, like this one: Directory Search Enter search criteria http://10.3.100.190:8080/ciscodirectory?action=list&page=0 First Name firstname A Last Name lastname A Number number T Copy the information from the URL line (http://10.3.100.190:8080/ciscodirectory?action=list&page=0). The easiest way to set the path in your phone is by the Web interface (but it could also be done by Telnet). Connect to your phone web server. Login and password are normally: admin Select Configure common phonebook. In Select phone book to use chose the value: Remote Then click on submit. File the box below with the URL you get previously: The IP address and the port number of the Phonebook server can be manually entered or synchronised with the Call Agent In our case, IP address: 10.3.100.190, Port number: 8080,Path: /ciscodirectory?action=list&page=0 Then click on submit. If you return to your phone and select the common phone book, it is normally connected to the remote one now. You can search by a name or if you put nothing and press on OK, it will return you the entire content of the remote phone book. More information about Cisco Phonebook management To manage remote phone book, you need a server. It could be one you developed by yourself or one include in your proxy server (not all of them include this feature). It must follow the Cisco implementation; you can have more information here: http://www.cisco.com/univercd/cc/td/doc/product/voice/vpdd/cdd/3_1/index.htm Check for "Cisco IP Phone Services Application Development Notes with Cisco CallManager 3.1." Igor Briski wrote: Anybody got any documentation/experience on the subject? I'm trying to get it working, but the documentation I have lacks any information on what should be installed on the server side. -- Igor Briški - [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP gateway: call hangups afer 3 rings
Hi all,
I've setup an asterisk gw (yyy.yyy.yyy.yyy), connected via IAX2 with another
asterisk (xxx.xxx.xxx.xxx) which acts as a centrex, and via SIP for
incoming/outgoing calls with a provider (zzz.zzz.zzz.zzz).
The problem is that outgoing calls are hanguped after three rings if they are
not answered, and from the debug trace, it seems to be the asterisk gw
who hangups. Apart from that, calls answered before three rings are handled
correctly.
I don't really see what could explain such comportement, and can't find a
"related" sip.conf parameter from the docs, or sample configs.
If anyone has an idea, I've included the related configs and the trace of a
call.
Best regards,
Nicolas Olivier
The gateway is running asterisk 1.0.7.
sip.conf:
[general]
context=default
port=5060
bindaddr=yyy.yyy.yyy.yyy
srvlookup=yes
[provider]
type=friend
host=zzz.zzz.zzz.zzz
port=5060
nat=yes
extensions.conf:
[default]
exten => _x.,1,Dial(SIP/[EMAIL PROTECTED])
exten => _x.,2,Hangup
exten => _x.,3,Congestion
(...)
Call debug:
-- Accepting AUTHENTICATED call from xxx.xxx.xxx.xxx requested format = 2,
actual format = 2
-- Executing Dial("IAX2/[EMAIL PROTECTED]/1", "SIP/[EMAIL PROTECTED]") in
new stack
We're at yyy.yyy.yyy.yyy port 12108
Answering/Requesting with root capability 0x2 (gsm)
Answering with capability 0x4 (ulaw)
Answering with capability 0x8 (alaw)
Answering with non-codec capability 0x1 (telephone-event)
12 headers, 12 lines
Reliably Transmitting:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bK5922a0f1;rport
From: "Choco Bobo" ;tag=as1a492e28
To:
Contact:
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Fri, 19 Sep 1980 10:09:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 265
v=0
o=root 13764 13764 IN IP4 yyy.yyy.yyy.yyy
s=session
c=IN IP4 yyy.yyy.yyy.yyy
t=0 0
m=audio 12108 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
(NAT) to zzz.zzz.zzz.zzz:5060
-- Called [EMAIL PROTECTED]
Sip read:
SIP/2.0 100 Trying
Allow: UPDATE,REFER
Call-ID: [EMAIL PROTECTED]
Contact:
CSeq: 102 INVITE
From: "Choco Bobo" ;tag=as1a492e28
Server: Cirpack/v4.38f (gw_sip)
To:
Via: SIP/2.0/UDP
yyy.yyy.yyy.yyy:5060;received=yyy.yyy.yyy.yyy;rport=5060;branch=z9hG4bK5922a0f1
Content-Length: 0
10 headers, 0 lines
Sip read:
SIP/2.0 183 In band info available
Allow: UPDATE,REFER
Call-ID: [EMAIL PROTECTED]
Contact:
Content-Type: application/sdp
CSeq: 102 INVITE
From: "Choco Bobo" ;tag=as1a492e28
Server: Cirpack/v4.38f (gw_sip)
To: ;tag=01-08086-78a18de8-67bc990a2
Via: SIP/2.0/UDP
yyy.yyy.yyy.yyy:5060;received=yyy.yyy.yyy.yyy;rport=5060;branch=z9hG4bK5922a0f1
Content-Length: 201
v=0
o=cp10 112987934014 112987934014 IN IP4 bbb.bbb.bbb.bbb
s=SIP Call
c=IN IP4 aaa.aaa.aaa.aaa
t=0 0
m=audio 30772 RTP/AVP 0 8
b=AS:64
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=ptime:20
11 headers, 10 lines
Found RTP audio format 0
Found RTP audio format 8
Peer audio RTP is at port aaa.aaa.aaa.aaa:30772
Found description format PCMU
Found description format PCMA
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xc
(ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities: us - 0x1 (g723), peer - 0x0 (nothing), combined - 0x0
(nothing)
Non-codec capabilities: us - 0x1 (g723), peer - 0x0 (nothing), combined - 0x0
(nothing)
-- SIP/b3g-7bfa is making progress passing it to IAX2/[EMAIL PROTECTED]/1
Reliably Transmitting:
CANCEL sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bK5922a0f1;rport
From: "Choco Bobo" ;tag=as1a492e28
To:
Contact:
Call-ID: [EMAIL PROTECTED]
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Content-Length: 0
(NAT) to zzz.zzz.zzz.zzz:5060
Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms
== Spawn extension (default, 0123456789, 4) exited non-zero on 'IAX2/[EMAIL
PROTECTED]/1'
-- Hungup 'IAX2/[EMAIL PROTECTED]/1'
Sip read:
SIP/2.0 200 OK
Call-ID: [EMAIL PROTECTED]
CSeq: 102 CANCEL
From: "Choco Bobo" ;tag=as1a492e28
To:
Via: SIP/2.0/UDP
yyy.yyy.yyy.yyy:5060;received=yyy.yyy.yyy.yyy;rport=5060;branch=z9hG4bK5922a0f1
Content-Length: 0
(...)
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[Asterisk-Users] SIP gateway: call hangups afer 3 rings
Hi all,
I've setup an asterisk gw (yyy.yyy.yyy.yyy), connected via IAX2 with another asterisk (xxx.xxx.xxx.xxx) which acts as a centrex, and via SIP for
incoming/outgoing calls with a provider (zzz.zzz.zzz.zzz).
The problem is that outgoing calls are hanguped after three rings if they are not answered, and from the debug trace, it seems to be the asterisk gw
who hangups. Apart from that, calls answered before three rings are handled correctly.
I don't really see what could explain such comportement, and can't find a
"related" sip.conf parameter from the docs, or sample configs.
If anyone has an idea, I've included the related configs and the trace of a
call.
Best regards,
Nicolas Olivier
The gateway is running asterisk 1.0.7.
sip.conf:
[general]
context=default
port=5060
bindaddr=yyy.yyy.yyy.yyy
srvlookup=yes
[provider]
type=friend
host=zzz.zzz.zzz.zzz
port=5060
nat=yes
extensions.conf:
[default]
exten => _x.,1,Dial(SIP/[EMAIL PROTECTED])
exten => _x.,2,Hangup
exten => _x.,3,Congestion
(...)
Call debug:
-- Accepting AUTHENTICATED call from xxx.xxx.xxx.xxx requested format = 2,
actual format = 2
-- Executing Dial("IAX2/[EMAIL PROTECTED]/1", "SIP/[EMAIL PROTECTED]") in
new stack
We're at yyy.yyy.yyy.yyy port 12108
Answering/Requesting with root capability 0x2 (gsm)
Answering with capability 0x4 (ulaw)
Answering with capability 0x8 (alaw)
Answering with non-codec capability 0x1 (telephone-event)
12 headers, 12 lines
Reliably Transmitting:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bK5922a0f1;rport
From: "Choco Bobo" ;tag=as1a492e28
To:
Contact:
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Fri, 19 Sep 1980 10:09:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 265
v=0
o=root 13764 13764 IN IP4 yyy.yyy.yyy.yyy
s=session
c=IN IP4 yyy.yyy.yyy.yyy
t=0 0
m=audio 12108 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
(NAT) to zzz.zzz.zzz.zzz:5060
-- Called [EMAIL PROTECTED]
Sip read:
SIP/2.0 100 Trying
Allow: UPDATE,REFER
Call-ID: [EMAIL PROTECTED]
Contact:
CSeq: 102 INVITE
From: "Choco Bobo" ;tag=as1a492e28
Server: Cirpack/v4.38f (gw_sip)
To:
Via: SIP/2.0/UDP
yyy.yyy.yyy.yyy:5060;received=yyy.yyy.yyy.yyy;rport=5060;branch=z9hG4bK5922a0f1
Content-Length: 0
10 headers, 0 lines
Sip read:
SIP/2.0 183 In band info available
Allow: UPDATE,REFER
Call-ID: [EMAIL PROTECTED]
Contact:
Content-Type: application/sdp
CSeq: 102 INVITE
From: "Choco Bobo" ;tag=as1a492e28
Server: Cirpack/v4.38f (gw_sip)
To: ;tag=01-08086-78a18de8-67bc990a2
Via: SIP/2.0/UDP
yyy.yyy.yyy.yyy:5060;received=yyy.yyy.yyy.yyy;rport=5060;branch=z9hG4bK5922a0f1
Content-Length: 201
v=0
o=cp10 112987934014 112987934014 IN IP4 bbb.bbb.bbb.bbb
s=SIP Call
c=IN IP4 aaa.aaa.aaa.aaa
t=0 0
m=audio 30772 RTP/AVP 0 8
b=AS:64
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=ptime:20
11 headers, 10 lines
Found RTP audio format 0
Found RTP audio format 8
Peer audio RTP is at port aaa.aaa.aaa.aaa:30772
Found description format PCMU
Found description format PCMA
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xc
(ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities: us - 0x1 (g723), peer - 0x0 (nothing), combined - 0x0
(nothing)
Non-codec capabilities: us - 0x1 (g723), peer - 0x0 (nothing), combined - 0x0
(nothing)
-- SIP/b3g-7bfa is making progress passing it to IAX2/[EMAIL PROTECTED]/1
Reliably Transmitting:
CANCEL sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bK5922a0f1;rport
From: "Choco Bobo" ;tag=as1a492e28
To:
Contact:
Call-ID: [EMAIL PROTECTED]
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Content-Length: 0
(NAT) to zzz.zzz.zzz.zzz:5060
Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms
== Spawn extension (default, 0123456789, 4) exited non-zero on 'IAX2/[EMAIL
PROTECTED]/1'
-- Hungup 'IAX2/[EMAIL PROTECTED]/1'
Sip read:
SIP/2.0 200 OK
Call-ID: [EMAIL PROTECTED]
CSeq: 102 CANCEL
From: "Choco Bobo" ;tag=as1a492e28
To:
Via: SIP/2.0/UDP
yyy.yyy.yyy.yyy:5060;received=yyy.yyy.yyy.yyy;rport=5060;branch=z9hG4bK5922a0f1
Content-Length: 0
(...)
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Re: [Asterisk-Users] zaphfc troubles
Stuart,
I switched the system to a pentium based host, with different memory.
The results are the same. I've also changed the ISDN card to be sure.
Nicolas
Stuart Hirst wrote:
> Nicolas,
>
> I replied earlier stating that I saw similar issues and now that you
> have applied the Florz patch the symptoms you are seeing are all but
> identical to the issues I saw and resolved by changing out the
> motherboard memory. The system was an ASUS main board with a Xeon
> processor.
>
> It is not the memory it could be something specific to the VIA motherboard.
>
> Stuart
>
>
>
> Nicolas Olivier wrote:
>
>>Well, afer applying zaphfc_0.2.0-RC8a_florz-6.diff, I'm highly flooded
> after ztcfg with:
>>
>>May 18 18:11:33 gw-ss daemon.crit klogd: zaphfc[0]: b channel buffer
> underrun: 0, 0
>>May 18 18:11:33 gw-ss daemon.crit klogd: zaphfc[0]: b channel buffer
> overflow: 311, 311
>>May 18 18:11:33 gw-ss daemon.crit klogd: zaphfc[0]: b channel buffer
> overflow: 436, 436
>>May 18 18:11:33 gw-ss daemon.crit klogd: zaphfc[0]: b channel buffer
> underrun: 0, 0
>>
>>And when I start asterisk, same stuff, kernel crashes.
>>
>>Interrupts are ok.
>>
>>sjaak imap wrote:
>>
>>
>>>Dear Nicolas Olivier
>>>
>>>Just try the florz patch at http://zaphfc.florz.dyndns.org/
>>>and look at cat /proc/interupts if your not sharing irq's
>>>
>>>Maybe this will help
>>>
>>>
>>>Good luck
>>>
>>>Sjaak
>>>
>>>
>>>
>>>>Hi,
>>>>
>>>>I'm trying to setup a small BRI ISDN <-> voip gateway.
>>>>The ISDN card is based on Cologne chipset, so I try set it up with
> zaphfc.
>>>>
>>>>The versions i'm running:
>>>>kernel-2.4.27
>>>>Asterisk 1.0.7-BRIstuffed-0.2.0-RC8e
>>>>zaptel modules 1.0.7
>>>>zaphfc is from bristuff-0.2.0-RC8e
>>>>
>>>>When I'm doing the insmod on zaptel, zaphfc, zaprtc:
>>>>
>>>>Zapata Telephony Interface Registered on major 196
>>>>PCI: Found IRQ 12 for device 00:12.0
>>>>zaphfc: CCD/Billion/Asuscom 2BD0 configured at mem 0xc384 fifo
>>>>
>>>>
>>>0xc2d58000(0x2d58000) IRQ 12 HZ 100
>>>
>>>
>>>>zaphfc: Card 0 configured for TE mode
>>>>Registered Span 1 ('ZTHFC1') with 3 channels
>>>>Span ('ZTHFC1') is new master
>>>>zaphfc: 1 hfc-pci card(s) in this box.
>>>>Registered Span 2 ('ZTRTC/1') with 0 channels
>>>>Real Time Clock Driver v1.10e
>>>>
>>>>I'm using zaprtc as the gateway is running on a VIA motherboard without
>>>>
>>>>
>>>USB controller.
>>>
>>>
>>>>When I'm doing ztcfg -vv:
>>>>
>>>>Zaptel Configuration
>>>>==
>>>>
>>>>SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1)
>>>>
>>>>Channel map:
>>>>
>>>>Channel 01: Individual Clear channel (Default) (Slaves: 01)
>>>>Channel 02: Individual Clear channel (Default) (Slaves: 02)
>>>>Channel 03: D-channel (Default) (Slaves: 03)
>>>>
>>>>3 channels configured.
>>>>
>>>>Here are my confs:
>>>>
>>>>/etc/zaptel.conf:
>>>>
>>>>loadzone=fr
>>>>defaultzone=fr
>>>>
>>>>span=1,1,3,ccs,ami
>>>>bchan=1-2
>>>>dchan=3
>>>>
>>>>/etc/asterisk/zapata.conf:
>>>>
>>>>[channels]
>>>>
>>>>language=fr
>>>>context=test
>>>>switchtype=euroisdn
>>>>signalling=bri_cpe
>>>>echocancel=yes
>>>>immediate=yes
>>>>channel => 1-2
>>>>
>>>>/etc/asterisk/modules.conf:
>>>>
>>>>[modules]
>>>>autoload=yes
>>>>
>>>>noload => pbx_gtkconsole.so
>>>>noload => pbx_kdeconsole.so
>>>>
>>>>noload => app_intercom.so
>>>>
>>>>load => chan_modem.so
>>>>load => res_features.so
>>>>load => res_musiconhold.so
>>>>load => chan_zap.so
>>>>
>>>>noload => chan_alsa.so
>>>>noload => chan_oss.so
>>>>
>>>>[global]
>>>>chan_modem.so=yes
>>>>
Re: [Asterisk-Users] zaphfc troubles
Quoting from:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20Zaptel%20Installation
As I haven't got a Digium card, I need a timer which can be provided by
ztdummy, zaprtc or zaprai.
But anyway the results are the same with or without zaprtc loaded.
Peer Oliver Schmidt wrote:
> Nicolas Olivier wrote:
>
>> I'm trying to setup a small BRI ISDN <-> voip gateway.
>> The ISDN card is based on Cologne chipset, so I try set it up with
> zaphfc.
>>
>> The versions i'm running:
>> kernel-2.4.27
>> Asterisk 1.0.7-BRIstuffed-0.2.0-RC8e
>> zaptel modules 1.0.7
>> zaphfc is from bristuff-0.2.0-RC8e
>>
>> When I'm doing the insmod on zaptel, zaphfc, zaprtc:
>>
>> Zapata Telephony Interface Registered on major 196
>> PCI: Found IRQ 12 for device 00:12.0
>> zaphfc: CCD/Billion/Asuscom 2BD0 configured at mem 0xc384 fifo
> 0xc2d58000(0x2d58000) IRQ 12 HZ 100
>> zaphfc: Card 0 configured for TE mode
>> Registered Span 1 ('ZTHFC1') with 3 channels
>> Span ('ZTHFC1') is new master
>> zaphfc: 1 hfc-pci card(s) in this box.
>> Registered Span 2 ('ZTRTC/1') with 0 channels
>> Real Time Clock Driver v1.10e
>>
>> I'm using zaprtc as the gateway is running on a VIA motherboard
> without USB controller.
> [..]
>
> Why are you running zaprtc? zaphfc provides your needed timing source.
> --
> Best regards
>
> Peer Oliver Schmidt
> PGP Key ID: 0x83E1C2EA
>
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Re: [Asterisk-Users] zaphfc troubles
Well, afer applying zaphfc_0.2.0-RC8a_florz-6.diff, I'm highly flooded after
ztcfg with:
May 18 18:11:33 gw-ss daemon.crit klogd: zaphfc[0]: b channel buffer underrun:
0, 0
May 18 18:11:33 gw-ss daemon.crit klogd: zaphfc[0]: b channel buffer overflow:
311, 311
May 18 18:11:33 gw-ss daemon.crit klogd: zaphfc[0]: b channel buffer overflow:
436, 436
May 18 18:11:33 gw-ss daemon.crit klogd: zaphfc[0]: b channel buffer underrun:
0, 0
And when I start asterisk, same stuff, kernel crashes.
Interrupts are ok.
sjaak imap wrote:
> Dear Nicolas Olivier
>
> Just try the florz patch at http://zaphfc.florz.dyndns.org/
> and look at cat /proc/interupts if your not sharing irq's
>
> Maybe this will help
>
>
> Good luck
>
> Sjaak
>
>>Hi,
>>
>>I'm trying to setup a small BRI ISDN <-> voip gateway.
>>The ISDN card is based on Cologne chipset, so I try set it up with zaphfc.
>>
>>The versions i'm running:
>>kernel-2.4.27
>>Asterisk 1.0.7-BRIstuffed-0.2.0-RC8e
>>zaptel modules 1.0.7
>>zaphfc is from bristuff-0.2.0-RC8e
>>
>>When I'm doing the insmod on zaptel, zaphfc, zaprtc:
>>
>>Zapata Telephony Interface Registered on major 196
>>PCI: Found IRQ 12 for device 00:12.0
>>zaphfc: CCD/Billion/Asuscom 2BD0 configured at mem 0xc384 fifo
> 0xc2d58000(0x2d58000) IRQ 12 HZ 100
>>zaphfc: Card 0 configured for TE mode
>>Registered Span 1 ('ZTHFC1') with 3 channels
>>Span ('ZTHFC1') is new master
>>zaphfc: 1 hfc-pci card(s) in this box.
>>Registered Span 2 ('ZTRTC/1') with 0 channels
>>Real Time Clock Driver v1.10e
>>
>>I'm using zaprtc as the gateway is running on a VIA motherboard without
> USB controller.
>>
>>When I'm doing ztcfg -vv:
>>
>>Zaptel Configuration
>>==
>>
>>SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1)
>>
>>Channel map:
>>
>>Channel 01: Individual Clear channel (Default) (Slaves: 01)
>>Channel 02: Individual Clear channel (Default) (Slaves: 02)
>>Channel 03: D-channel (Default) (Slaves: 03)
>>
>>3 channels configured.
>>
>>Here are my confs:
>>
>>/etc/zaptel.conf:
>>
>>loadzone=fr
>>defaultzone=fr
>>
>>span=1,1,3,ccs,ami
>>bchan=1-2
>>dchan=3
>>
>>/etc/asterisk/zapata.conf:
>>
>>[channels]
>>
>>language=fr
>>context=test
>>switchtype=euroisdn
>>signalling=bri_cpe
>>echocancel=yes
>>immediate=yes
>>channel => 1-2
>>
>>/etc/asterisk/modules.conf:
>>
>>[modules]
>>autoload=yes
>>
>>noload => pbx_gtkconsole.so
>>noload => pbx_kdeconsole.so
>>
>>noload => app_intercom.so
>>
>>load => chan_modem.so
>>load => res_features.so
>>load => res_musiconhold.so
>>load => chan_zap.so
>>
>>noload => chan_alsa.so
>>noload => chan_oss.so
>>
>>[global]
>>chan_modem.so=yes
>>chan_zap.so=yes
>>
>>
>>The problem is that after ztcfg ran, I've got the following logs:
>>
>>Registered tone zone 2 (France)
>>zaphfc: card 0 layer 1 state = F4
>>zaphfc: card 0 layer 1 state = F5
>>zaphfc: card 0 layer 1 state = F7
>>zaphfc: card 0 RX [ 0xfe 0xff 0x3 0xf 0x0 0x0 0x4 0xff ] 8 bytes
>>zaphfc: card 0 RX [ 0xfe 0xff 0x3 0xf 0x0 0x0 0x4 0xff ] 8 bytes
>>zaphfc: card 0 layer 1 state = F3
>>zaphfc: card 0 layer 1 state = F4
>>zaphfc: card 0 layer 1 state = F5
>>zaphfc: card 0 layer 1 state = F7
>>zaphfc: bchan rx fifo not enough bytes to receive! (z1=5630, z2=5623,
> wanted 8 got 7), probably a buffer overrun.
>>zaphfc: bchan rx fifo not enough bytes to receive! (z1=6163, z2=6156,
> wanted 8 got 7), probably a buffer overrun.
>>zaphfc: card 0 RX [ 0xfe 0xff 0x3 0xf 0x0 0x0 0x4 0xff ] 8 bytes
>>zaphfc: card 0 RX [ 0xfe 0xff 0x3 0xf 0x0 0x0 0x4 0xff ] 8 bytes
>>
>>And when I start asterisk -c, same logs keep on, and I've finally a
> kernel crash:
>>
>>Unable to handle kernel paging request at virtual address fffc
>> printing eip:
>> c0113cc0
>> *pde = d063
>> *pte =
>> Oops:
>> CPU:0
>> EIP:0010:[]Not tainted
>> EFLAGS: 00010013
>> eax: c248015c ebx: ecx: 0001 edx: 0001
>> esi: c24803a0 edi: c248015c ebp: c2c8fe2c esp: c2c8fe14
>> ds: 0018 es: 0018 ss: 0018
>> Process sshd (pid: 146, stackpage=c2c8f000)
>> Stack: 0001 0086 0001 c24803a0 c24803a0 c270c940 c248
> c3
Re: [Asterisk-Users] zaphfc troubles
Just an update, I deoopsed the kernel dump, must be usable...
Nicolas Olivier wrote:
>
> Hi,
>
> I'm trying to setup a small BRI ISDN <-> voip gateway.
> The ISDN card is based on Cologne chipset, so I try set it up with zaphfc.
>
> The versions i'm running:
> kernel-2.4.27
> Asterisk 1.0.7-BRIstuffed-0.2.0-RC8e
> zaptel modules 1.0.7
> zaphfc is from bristuff-0.2.0-RC8e
>
> When I'm doing the insmod on zaptel, zaphfc, zaprtc:
>
> Zapata Telephony Interface Registered on major 196
> PCI: Found IRQ 12 for device 00:12.0
> zaphfc: CCD/Billion/Asuscom 2BD0 configured at mem 0xc384 fifo
> 0xc2d58000(0x2d58000) IRQ 12 HZ 100
> zaphfc: Card 0 configured for TE mode
> Registered Span 1 ('ZTHFC1') with 3 channels
> Span ('ZTHFC1') is new master
> zaphfc: 1 hfc-pci card(s) in this box.
> Registered Span 2 ('ZTRTC/1') with 0 channels
> Real Time Clock Driver v1.10e
>
> I'm using zaprtc as the gateway is running on a VIA motherboard without
> USB controller.
>
> When I'm doing ztcfg -vv:
>
> Zaptel Configuration
> ==
>
> SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1)
>
> Channel map:
>
> Channel 01: Individual Clear channel (Default) (Slaves: 01)
> Channel 02: Individual Clear channel (Default) (Slaves: 02)
> Channel 03: D-channel (Default) (Slaves: 03)
>
> 3 channels configured.
>
> Here are my confs:
>
> /etc/zaptel.conf:
>
> loadzone=fr
> defaultzone=fr
>
> span=1,1,3,ccs,ami
> bchan=1-2
> dchan=3
>
> /etc/asterisk/zapata.conf:
>
> [channels]
>
> language=fr
> context=test
> switchtype=euroisdn
> signalling=bri_cpe
> echocancel=yes
> immediate=yes
> channel => 1-2
>
> /etc/asterisk/modules.conf:
>
> [modules]
> autoload=yes
>
> noload => pbx_gtkconsole.so
> noload => pbx_kdeconsole.so
>
> noload => app_intercom.so
>
> load => chan_modem.so
> load => res_features.so
> load => res_musiconhold.so
> load => chan_zap.so
>
> noload => chan_alsa.so
> noload => chan_oss.so
>
> [global]
> chan_modem.so=yes
> chan_zap.so=yes
>
>
> The problem is that after ztcfg ran, I've got the following logs:
>
> Registered tone zone 2 (France)
> zaphfc: card 0 layer 1 state = F4
> zaphfc: card 0 layer 1 state = F5
> zaphfc: card 0 layer 1 state = F7
> zaphfc: card 0 RX [ 0xfe 0xff 0x3 0xf 0x0 0x0 0x4 0xff ] 8 bytes
> zaphfc: card 0 RX [ 0xfe 0xff 0x3 0xf 0x0 0x0 0x4 0xff ] 8 bytes
> zaphfc: card 0 layer 1 state = F3
> zaphfc: card 0 layer 1 state = F4
> zaphfc: card 0 layer 1 state = F5
> zaphfc: card 0 layer 1 state = F7
> zaphfc: bchan rx fifo not enough bytes to receive! (z1=5630, z2=5623,
> wanted 8 got 7), probably a buffer overrun.
> zaphfc: bchan rx fifo not enough bytes to receive! (z1=6163, z2=6156,
> wanted 8 got 7), probably a buffer overrun.
> zaphfc: card 0 RX [ 0xfe 0xff 0x3 0xf 0x0 0x0 0x4 0xff ] 8 bytes
> zaphfc: card 0 RX [ 0xfe 0xff 0x3 0xf 0x0 0x0 0x4 0xff ] 8 bytes
>
> And when I start asterisk -c, same logs keep on, and I've finally a
> kernel crash:
>
> Unable to handle kernel paging request at virtual address fffc
> printing eip:
> c0113cc0
> *pde = d063
> *pte =
> Oops:
> CPU:0
> EIP:0010:[]Not tainted
> EFLAGS: 00010013
> eax: c248015c ebx: ecx: 0001 edx: 0001
> esi: c24803a0 edi: c248015c ebp: c2c8fe2c esp: c2c8fe14
> ds: 0018 es: 0018 ss: 0018
> Process sshd (pid: 146, stackpage=c2c8f000)
> Stack: 0001 0086 0001 c24803a0 c24803a0 c270c940 c248
> c3819545
> 0010 0010 c2c8ff24 0046 1140 0003 c2c8ffc4
> 0086
> c01cb6b1 c02f8bc4 c24803a0 8005003b c2c8feb4 0002 0008
> c270c800
> Call Trace:[] [] [] []
> []
> [] [] [] [] []
> []
> []
>
>>EIP; c0113cc0 <__wake_up+20/a0> <=
>>eax; c248015c <_end+217f0d0/34fef74>
>>esi; c24803a0 <_end+217f314/34fef74>
>>edi; c248015c <_end+217f0d0/34fef74>
>>ebp; c2c8fe2c <_end+298eda0/34fef74>
>>esp; c2c8fe14 <_end+298ed88/34fef74>
Trace; c3819545 <[zaptel]__zt_receive_chunk+133d/1484>
Trace; c01cb6b1 <__ide_do_rw_disk+3e1/650>
Trace; c381aae6 <[zaptel]zt_receive+a26/b0c>
Trace; c381aad7 <[zaptel]zt_receive+a17/b0c>
Trace; c383cd78 <[zaphfc]hfc_interrupt+228/358>
Trace; c01cae16
Trace; c383ce95 <[zaphfc]hfc_interrupt+345/358>
Trace; c01c5416
Trace; c01cad01
Trace; c0109ddd
Trace; c0109f78
Trace; c010c328
Code; c0113cc0 <__wake_up+20/a0>
[Asterisk-Users] zaphfc troubles
038 == Parsing '/etc/asterisk/rtp.conf': Found == RTP Allocating from port range 1 -> 2 Asterisk PBX Core Initializing Registering builtin applications: [AbsoluteTimeout] == Registered application 'AbsoluteTimeout' (...) [WaitExten] == Registered application 'WaitExten' Asterisk Dynamic Loader Starting: == Parsing '/etc/asterisk/modules.conf': Found [chan_modem.so] => (Generic Voice Modem Driver) == Parsing '/etc/asterisk/modem.conf': Found == Loading modem driver chan_modem_aopen.so => (A/Open (Rockwell Chipset) ITU-2 VoiceModem Driver) == Registered channel type 'Modem' (Generic Voice Modem Channel Driver) [res_features.so] => (Call Parking Resource) == Parsing '/etc/asterisk/features.conf': Found -- Registered extension context 'parkedcalls' -- Added extension '700' priority 1 to parkedcalls == Registered application 'ParkedCall' == Registered application 'Park' == Manager registered action ParkedCalls == Registered application 'HoldedCall' == Registered application 'AutoanswerLogin' == Registered application 'Autoanswer' [res_musiconhold.so] => (Music On Hold Resource) == Parsing '/etc/asterisk/musiconhold.conf': Found == Registered application 'MusicOnHold' == Registered application 'WaitMusicOnHold' == Registered application 'SetMusicOnHold' [chan_zap.so] The ISDN line has been validated, and the ISDN is known to work. I've searched in the archives, wiki, and can't see what's wrong. If anyone has an advice, it will be greatly appreciated. Nicolas Olivier ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
