[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 sip:[EMAIL PROTECTED];tag=as1a492e28 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] 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: sip:zzz.zzz.zzz.zzz:5060 CSeq: 102 INVITE From: Choco Bobo sip:[EMAIL PROTECTED];tag=as1a492e28 Server: Cirpack/v4.38f (gw_sip) To: sip:[EMAIL PROTECTED] 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: sip:zzz.zzz.zzz.zzz:5060 Content-Type: application/sdp CSeq: 102 INVITE From: Choco Bobo sip:[EMAIL PROTECTED];tag=as1a492e28 Server: Cirpack/v4.38f (gw_sip) To: sip:[EMAIL PROTECTED];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 sip:[EMAIL PROTECTED];tag=as1a492e28 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] 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 sip:[EMAIL PROTECTED];tag=as1a492e28 To: sip:[EMAIL PROTECTED] Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;received=yyy.yyy.yyy.yyy;rport=5060;branch=z9hG4bK5922a0f1 Content-Length: 0 (...) ___ --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 sip:[EMAIL PROTECTED];tag=as1a492e28 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] 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: sip:zzz.zzz.zzz.zzz:5060 CSeq: 102 INVITE From: Choco Bobo sip:[EMAIL PROTECTED];tag=as1a492e28 Server: Cirpack/v4.38f (gw_sip) To: sip:[EMAIL PROTECTED] 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: sip:zzz.zzz.zzz.zzz:5060 Content-Type: application/sdp CSeq: 102 INVITE From: Choco Bobo sip:[EMAIL PROTECTED];tag=as1a492e28 Server: Cirpack/v4.38f (gw_sip) To: sip:[EMAIL PROTECTED];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 sip:[EMAIL PROTECTED];tag=as1a492e28 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] 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 sip:[EMAIL PROTECTED];tag=as1a492e28 To: sip:[EMAIL PROTECTED] Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;received=yyy.yyy.yyy.yyy;rport=5060;branch=z9hG4bK5922a0f1 Content-Length: 0 (...) ___ --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
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: CiscoIPPhoneInput TitleDirectory Search/Title PromptEnter search criteria/Prompt URLhttp://10.3.100.190:8080/ciscodirectory?action=listpage=0/URL InputItem DisplayNameFirst Name/DisplayName QueryStringParamfirstname/QueryStringParam InputFlagsA/InputFlags /InputItem InputItem DisplayNameLast Name/DisplayName QueryStringParamlastname/QueryStringParam InputFlagsA/InputFlags /InputItem InputItem DisplayNameNumber/DisplayName QueryStringParamnumber/QueryStringParam InputFlagsT/InputFlags /InputItem /CiscoIPPhoneInput Copy the information from the URL line (URLhttp://10.3.100.190:8080/ciscodirectory?action=listpage=0/URL). 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=listpage=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] zaphfc troubles
- 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
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:[c0113cc0]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:[c3819545] [c01cb6b1] [c381aae6] [c381aad7]
[c383cd78]
[c01cae16] [c383ce95] [c01c5416] [c01cad01] [c0109ddd]
[c0109f78]
[c010c328]
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 read_intr+76/1b0
Trace; c383ce95 [zaphfc]hfc_interrupt+345/358
Trace; c01c5416 ide_intr+96/100
Trace; c01cad01 lba_capacity_is_ok+81/120
Trace; c0109ddd handle_IRQ_event+3d/70
Trace; c0109f78 do_IRQ+68/a0
Trace; c010c328 call_do_IRQ+5/d
Code; c0113cc0 __wake_up+20/a0
_EIP:
Code; c0113cc0 __wake_up+20/a0 =
0: 8b 4b fc mov0xfffc(%ebx),%ecx =
Code; c0113cc3 __wake_up+23/a0
3: 8b 01 mov(%ecx),%eax
Code; c0113cc5 __wake_up+25/a0
5: 85 45 f0 test %eax,0xfff0(%ebp)
Code; c0113cc8 __wake_up+28/a0
8: 74 56 je 60 _EIP+0x60 c0113d20
__wake_up+80/a0
Code; c0113cca __wake_up+2a/a0
a: 31 c0 xor%eax,%eax
Code; c0113ccc __wake_up+2c/a0
c: 9cpushf
Code; c0113ccd __wake_up+2d/a0
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:[c0113cc0]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:[c3819545] [c01cb6b1] [c381aae6] [c381aad7]
[c383cd78]
[c01cae16] [c383ce95] [c01c5416] [c01cad01] [c0109ddd]
[c0109f78]
[c010c328]
Code: 8b 4b fc 8b 01 85 45 f0 74 56 31 c0 9c 5e fa 8b 51 3c c7 01
0Kernel panic: Aiee, killing interrupt handler!
In interrupt handler - not syncing
Here is the output from asterisk:
No entry for terminal type screen;
using dumb terminal settings.
== Parsing '/etc/asterisk/asterisk.conf': Found
== Parsing '/etc/asterisk/extconfig.conf': Found
Asterisk 1.0.7-BRIstuffed-0.2.0-RC8e, Copyright (C) 1999-2004 Digium.
Written by Mark Spencer [EMAIL PROTECTED]
=
== Parsing '/etc/asterisk/logger.conf': Found
Asterisk Event Logger Started /var/log/asterisk/event_log
== Manager registered action Ping
== Manager registered action Events
== Manager registered action Logoff
== Manager registered action Hangup
== Manager registered
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
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
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:[c0113cc0]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:[c3819545] [c01cb6b1] [c381aae6] [c381aad7]
[c383cd78]
[c01cae16] [c383ce95] [c01c5416] [c01cad01] [c0109ddd]
[c0109f78]
[c010c328]
Code: 8b 4b fc 8b 01 85 45 f0 74 56 31 c0 9c 5e fa 8b 51 3c c7 01
0Kernel panic: Aiee, killing interrupt handler!
In interrupt handler
