Re: [asterisk-users] PRI (Primary-NTT)
On 1/7/2013 9:22 AM, Dan Austin wrote: The last PRI I setup in Hong Kong was configured as a Primary-Net5, which maps to euroisdn in DAHDI. That was eight years ago, so things may have changed, but it is worth a try. You should also collect some Q.931 logs, as I have seen silly things like caller-id formatting cause calls to be rejected. Sadly I cannot tell you how to accomplish that with DAHDI... we replaced the T1 card. turned out it's the hardware issue. now it's all fine. thanks. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PRI (Primary-NTT)
hi folks. i recently setup an Asterisk system in Hong Kong. their phone company told me that their T1 PRI switch type is Primary-NTT. however in chan_dahdi.conf there's no such option. i have it set to national. it worked fine for a while, but now suddenly stop working. in coming call just keep ringing and didn't even show up on console. out going call hang up immediately with cause code 27. (as usual, phone co. just said it's problem with our equipment without giving us any detail). anybody have any suggestions? here's our /etc/dahdi/system.conf: loadzone=hk defaultzone=hk span=1,1,0,esf,b8zs bchan=1-23 dchan=24 span=2,0,0,esf,b8zs bchan=25-47 dchan=48 span=3,0,0,esf,b8zs bchan=49-71 dchan=72 span=4,0,0,esf,b8zs bchan=73-95 dchan=96 /etc/asterisk/chan_dahdi.conf: switchtype=national pridialplan=unknown prilocaldialplan=unknown internationalprefix = 001 nationalprefix = unknownprefix = signalling=pri_cpe usecallerid=yes usecallingpres=yes echocancel=no echocancelwhenbridged=no group=1 callgroup=1 pickupgroup=1 faxdetect=incoming context=pri_in channel = 1-23 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI got event HDLC Abort
:315499683708 Function call interrupts TLB: 26154 53799 41831 28303 TLB shootdowns TRM: 0 0 0 0 Thermal event interrupts THR: 0 0 0 0 Threshold APIC interrupts MCE: 0 0 0 0 Machine check exceptions MCP: 2068 2068 2068 2068 Machine check polls ERR: 0 MIS: 0 -- Edwin Lam edwin@officegeneral.com Systems Engineer, OfficeWyze, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI got event HDLC Abort
On 11/2/2012 10:06 PM, Liban Abdi wrote: is there static on the line?? no. there were customer complains about sound cutting in and out. however i wasn't noticing and bad sound quality when i was testing it. is there timing slips and crc4 errors? no. the only messages i have are the HDLC abort warning. are they increasing throughout the day? they happen randomly, and quite frequently. are you getting timing slips during the day when users are using the phones and not off-peak hours? no timing slips related messages in either Asterisk's logs or syslog. are you getting hdlc abort erros when you hear a static noises?? that i don't know. however there was once it happened while i was in the middle of a call but i couldn't hear any sound drop off or any static. is the card sharing irq? no. this the only card that uses IRQ 30 1b:00.0 Network controller: Digium, Inc. Device 1420 (rev 14) Subsystem: Device 0005: Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV+ VGASnoop- ParErr+ Stepping- SERR+ FastB2B- DisINTx- Status: Cap- 66MHz- UDF- FastB2B- ParErr- DEVSEL=slow TAbort- TAbort- MAbort- SERR- PERR- INTx- Latency: 64 (8000ns min, 32000ns max), Cache Line Size: 64 bytes Interrupt: pin A routed to IRQ 30 Region 0: Memory at 97a0 (32-bit, non-prefetchable) [size=32K] Kernel driver in use: wct4xxp is your system plugged directly into an outlet without ups? good question. i don't know. On Fri, Nov 2, 2012 at 8:40 PM, Edwin Lam edwin@officegeneral.com mailto:edwin@officegeneral.com wrote: hi folks. recently some of our customers complained about bad voice quality on the phone system. i looked at the logs and found a lot of these: [2012-11-03 08:26:38] NOTICE[11305] chan_dahdi.c: PRI got event: HDLC Abort (6) on D-channel of span 1 [2012-11-03 08:26:45] NOTICE[11305] chan_dahdi.c: PRI got event: HDLC Abort (6) on D-channel of span 1 [2012-11-03 08:26:54] NOTICE[11305] chan_dahdi.c: PRI got event: HDLC Abort (6) on D-channel of span 1 i upgraded Asterisk/dahdi/libpri. tried turn on/off echo canceller etc. nothing seems to help. call the phone company to check out the line (which they said it's working fine) any idea? do i have a hardware issue here? i've check syslog there was no dahdi errors. here's my system.conf: span=1,1,0,esf,b8zs bchan=1-23 dchan=24 span=2,0,0,esf,b8zs bchan=25-47 dchan=48 span=3,0,0,esf,b8zs bchan=49-71 dchan=72 span=4,0,0,esf,b8zs bchan=73-95 dchan=96 and here's my chan_dahdi.conf: [channels] switchtype=national pridialplan=unknown prilocaldialplan=unknown internationalprefix = 001 nationalprefix = unknownprefix = signalling=pri_cpe usecallerid=yes usecallingpres=yes echocancel=no echocancelwhenbridged=no group=1 callgroup=1 pickupgroup=1 faxdetect=incoming context=defaultspan1 channel = 1-23 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PRI got event HDLC Abort
hi folks. recently some of our customers complained about bad voice quality on the phone system. i looked at the logs and found a lot of these: [2012-11-03 08:26:38] NOTICE[11305] chan_dahdi.c: PRI got event: HDLC Abort (6) on D-channel of span 1 [2012-11-03 08:26:45] NOTICE[11305] chan_dahdi.c: PRI got event: HDLC Abort (6) on D-channel of span 1 [2012-11-03 08:26:54] NOTICE[11305] chan_dahdi.c: PRI got event: HDLC Abort (6) on D-channel of span 1 i upgraded Asterisk/dahdi/libpri. tried turn on/off echo canceller etc. nothing seems to help. call the phone company to check out the line (which they said it's working fine) any idea? do i have a hardware issue here? i've check syslog there was no dahdi errors. here's my system.conf: span=1,1,0,esf,b8zs bchan=1-23 dchan=24 span=2,0,0,esf,b8zs bchan=25-47 dchan=48 span=3,0,0,esf,b8zs bchan=49-71 dchan=72 span=4,0,0,esf,b8zs bchan=73-95 dchan=96 and here's my chan_dahdi.conf: [channels] switchtype=national pridialplan=unknown prilocaldialplan=unknown internationalprefix = 001 nationalprefix = unknownprefix = signalling=pri_cpe usecallerid=yes usecallingpres=yes echocancel=no echocancelwhenbridged=no group=1 callgroup=1 pickupgroup=1 faxdetect=incoming context=defaultspan1 channel = 1-23 -- Edwin Lam edwin@officegeneral.com Systems Engineer, OfficeWyze, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Deleting OLD Voicemails
On 5/23/12 2:42 AM, Danny Dias wrote: Can i delete like this: rm -rf /var/spool/asterisk/voicemail/voicemailcontextcustomer/300/INBOX/*.* Is that ok? will this break something? that's ok no it shouldn't break anything. however if you're going to delete some of the messages. you have to renumber all the messages so that they are consecutive otherwise the voicemail application may give you grief. A little doubt here, once the user hears the voicemail using the phone, the message is automatically moved to Old folder, is that right? yes -- Edwin Lam edwin@officegeneral.com Systems Engineer, OfficeWyze, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime Registration
On 12/10/11 9:54 PM, Takehiro Matsushima wrote: I'd configured realtime registration, but configuration was not applied when I changed a row of sippeers table. To apply, 'sip reload' was needed (in Asterisk 1.8.0). or you can 'sip prune realtime extension' (On 12/08/2011 03:42), Andrew O. Zhukov wrote: No secrets :) SELECT * FROM sippeers WHERE name = '105680' AND host = 'dynamic' name|type|username|secret|fromuser|fromdomain|nat|context|canreinvite|disallow|allow|host|insecure|port|ipaddr|outboundproxy 105680|peer|testbutton2|XXX||button.ipshka.com:5060|no|button|no|all|speex;ulaw;alaw;g729;ilbc;g726;g726aal2;slin;lpc10;adpcm;g723|dynamic|port,invite|5060 ||ipshka.com On 12/07/2011 08:04 PM, Jonathan Rose wrote: [Dec 7 19:31:18] ERROR[4217]: chan_sip.c:9940 register_verify: Peer '105680' is trying to register, but not configured as host=dynamic Going on this, I'd say you probably tried to specify the host with a static IP address or a host name. If that's the case, you can't register, because that would be against the whole point of registering in the first place. You should probably post the DB entry for this peer to this thread to make things simpler... if it doesn't contain sensitive data. Of course, you can censor that out too. - Original Message - From: Andrew O. Zhukovgn...@telegroup.com.ua To: asterisk-users@lists.digium.com Sent: Wednesday, December 7, 2011 11:56:20 AM Subject: [asterisk-users] Realtime Registration [Dec 7 19:31:18] DEBUG[4217]: res_config_pgsql.c:821 pgsql_reconnect: Postgresql RealTime: Everything is fine. [Dec 7 19:31:18] DEBUG[4217]: res_config_pgsql.c:201 realtime_pgsql: Postgresql RealTime: Result=0x12e4edd0 Query: SELECT * FROM sippeers WHERE name = '105680' AND host = 'dynamic' [Dec 7 19:31:18] DEBUG[4217]: res_config_pgsql.c:209 realtime_pgsql: Postgresql RealTime: Found 1 rows. [Dec 7 19:31:18] ERROR[4217]: chan_sip.c:9940 register_verify: Peer '105680' is trying to register, but not configured as host=dynamic Any suggestions??? Asterisk 1.4.42 -- Edwin Lam edwin@officegeneral.com Systems Engineer, OfficeWyze, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] rasterisk not knowing config path?
On 12/6/11 9:18 AM, Yaroslav Panych wrote: Asterisk=1.8 No, I want manually connect to asterisk via ssh console. I.e. like: ssh user@ast-host /sbin/rasterisk -switch_to_point_asterisk.ctl path_to_asterisk.ctl I knowpath_to_asterisk.ctl but did not found any switch_to_point_asterisk.ctl in manuals. /usr/sbin/asterisk -r -s path_to_asterisk.ctl and you'll need both rw permission to that socket. (and minimum of x permission for the directory tree it resides in) -- Edwin Lam edwin@officegeneral.com Systems Engineer, OfficeWyze, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] skype connect early media
hi folks. when i use regular PSTN(sip phone - asterisk - PRI) to call certain numbers and when that number is unavailable. i usually hear an early media message saying blahblah is unavailable, please try again. but when i use skype connect(sip phone - asterisk - skype connect). i just hear ring back tone for about 20 seconds and then become fast busy. is there any setting i'm unaware of when setting up sip w/ skype connect? any suggestions would be appreciated. -- Edwin Lam edwin@officegeneral.com Systems Engineer, OfficeWyze, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to use password file with AuthenticateApplication
On 11/24/11 2:13 AM, virendra bhati wrote: I did the same as you mention like that *echo -n 1234 | md5sum another things to check are: - the permission of the file /tmp/pass.txt? - the user/group of that file? - the user/group asterisk running as? i'm using 1.6.2.18 and it works perfectly. maybe you can upgrade yours -- Edwin Lam edwin@officegeneral.com Systems Engineer, OfficeWyze, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to use password file with AuthenticateApplication
On 11/22/11 9:02 PM, virendra bhati wrote: On Mon, Nov 21, 2011 at 6:15 AM, virendra bhati virbh...@gmail.com mailto:virbh...@gmail.com wrote: Hi, After deleting all space no improvements. Try reversing the account code and password hash, like this: 81dc9bdb52d04dc20036dbd8313ed055:Virendra 9996535e07258a7bbfd8b132435c5962:Vijay 7bccfde7714a1ebadf06c5f4cea752c1:VirendraBhati the original format (i.e. accountcode:md5hash) was correct. one question, when you create the md5hash did you use the echo command? if so, did you specify the -n option? e.g. echo -n 12345 | md5sum -- Edwin Lam edwin@officegeneral.com Systems Engineer, OfficeWyze, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax not detected by Asterisk
On 11/17/11 3:30 AM, ik wrote: I have an Elastix 2 machine with digium fax modules (with license). When I try to create an extension that also works with FAX, Asterisk does not detect any incoming fax. Even when I use 'fax set debug', it does not display anything. It's Asterisk 6.2.x . Any ideas what can I do to figure out why it does not detect the arriving faxes ? in your dialplan, did you Answer the call and Wait a few seconds for Asterisk to detect the fax tone before you do some other things? -- Edwin Lam edwin@officegeneral.com Systems Engineer, OfficeWyze, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Binding asterisk to two static IPs
On 10/12/11 2:27 PM, ge...@riseup.net wrote: If you put 0.0.0.0, it will bind to all addresses. In a HA Cluster, on the active node, if you have a box address of 192.168.1.101 and a floating address of 192.168.1.102, then if you use bindaddr=0.0.0.0 you will find that phones on the 192.168.1.x subnet will not register on the floating address, which of course defeats the point of HA clustering. What happens is that the registration packets go to the floating address 192.168.1.102 but the response packets appear to come from 192.168.1.101 [same NIC but the packet contains the base address attached to the NIC], so the registration fails. Any idea how to solve this? try use 2 different subnet addresses instead of both addresses on the same subnet. e.g. 192.168.1.101/24 192.168.2.101/24 and also use ip command to add the address to the interface instead of ifconfig and eth0:x notation. that way the OS will pick the correct address when responding to in coming packets. the problem is when you assign 2 addresses to the same interface on the same subnet, one of them will be primary and the other will becomes secondary on that subnet. the OS will always pick the primary address when sending out packets on that subnet. -- Edwin Lam edwin@officegeneral.com Systems Engineer, OfficeWyze, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multiple Asterisk Sessions on same machine
On 7/19/11 11:07 AM, Michael wrote: We would like Asterisk to listen on port 5060 and on an additional port. From what we read online, it's not really possible, so is it possible to install a separate instance of Asterisk on the same machine (without using Vmware or such) and set the 2nd instance to listen on another port? all you have to do is setup port forwarding from an alternate port to 5060. we're doing that on our server. works like a charm. e.g. iptables -t nat -I PREROUTING 1 -i your-interface -d your-ip-addr \ -p udp -m udp --dport 6060 -j DNAT --to-destination your-ip-addr:5060 this sets up port 6060 to forward to 5060. -- Edwin Lam edwin@officegeneral.com Systems Engineer, OfficeWyze, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Different callerid for different extensions
On 6/7/11 1:59 AM, mahesh katta wrote:
I have small confusion in my configuration which is I had some DID's like
044578900-04457999. I was configured dial plan below mention.
you have 9 digits on the starting number 8 digits
on the ending number. i'll assume it's a typo and
the ending number is 04457 (total of 1100 DIDs)
exten = _0X,1,NoOp(Int exten:${CALLERID(num)})
exten = _0X,2,Set(outgoing_ident=0445789${CALLERID(num):-2})
exten = _0X,3,NoOp(Ext ident:${outgoing_ident})
exten = _0X,4,Set(CALLERID(name)=${outgoing_ident})
exten = _0X,5,AGI(agi://127.0.0.1:4577/call_log
http://127.0.0.1:4577/call_log)
exten = _0X,6,Set(CALLERID(num)=${outgoing_ident})
exten =
_0X,7,MixMonitor(/var/spool/asterisk/astrec/${TIMESTAMP}-${CALLERIDNUM}-${EXTEN}-${UNIQUEID}.gsm|av(0)V(0))
exten = _0X,8,Dial(${TRUNK}/${EXTEN},,tTo)
exten = _0X,9,Hangup
this dial plan for outbound .But I have some extensions that is like
100-110,200-210,300-310, etc. with this dialplan when I dial from 100 extension
callerid will show 044578900 right, and i have 200 extension also when i dial
from
this callerid will show 044578900 right. but i need to difine every extension
should be show different callerid . and same as INbound also.
Please anybody give me short dialplan for this .
change prio 2 line to the following:
exten =
_0X,2,Set(outgoing_ident=04457${MATH(8900+${MATH(${CALLERID(num)}-100)})})
i just did this out of my head, i haven't test it.
but this should map all 100-399 extensions to DID 044578900-0444579199
--
Edwin Lam edwin@officegeneral.com
Systems Engineer, OfficeWyze, Inc.
Ph: +1 415 439 4988 Fax: +1 415 283 3370
http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20
--
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Re: [asterisk-users] DAHDI span timeing source
On 5/27/11 2:20 PM, satish patel wrote: Tell me in one word. We have 2 PRI line connected with sangoma card what option would be good for me? 0 or 1 ? that would depends on what's the other end of the 2 PRI connected to. Date: Fri, 27 May 2011 16:11:03 -0500 From: sruff...@digium.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] DAHDI span timeing source On Fri, May 27, 2011 at 08:57:15PM +, satish patel wrote: You mean say 0=Slave (Use PSTN clock) 1=Master(generate Internal clock) So best option is 0 for all span if you connected on PSTN right ? Not really. Looking in system.conf.sample in dahdi-tools [1] Choose 1 to make the equipment at the far end of the E1/T1/BRI link the preferred source of the master clock. Choose 2 to make it the second choice for the master clock, if the first choice port fails (the far end dies, a cable breaks, or whatever). Choose 3 to make a port the third choice, and so on. If you have, say, 2 ports connected to the PSTN, mark those as 1 and 2. The number used for each port should be different. If you choose 0, the port will never be used as a source of timing. This is appropriate when you know the far end should always be a slave to you. If the port is connected to a channel bank, for example, you should always be its master. Likewise, BRI TE ports should always be configured as a slave. Any number of ports can be marked as 0. [1] http://svn.asterisk.org/view/dahdi/tools/trunk/system.conf.sample?view=co -- Edwin Lam edwin@officegeneral.com Systems Engineer, OfficeWyze, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6: Custom Name for Recordings file
On 5/13/11 10:57 AM, RSCL Mumbai wrote:
I have latest Elastix 64 bit setup and running fine (Asterisk
1.6.2.13)
I would like to customize the file name of call recordings:
/var/spool/asterisk/monitor/20110511-110858-1305126538.912.wav
I would like to include the extension number in the file name.
Did a lot of googling but not much help.
Pls advice.
[snip..]
Thx Eric.
I read the link e1*CLI core show application monitor but I could not follow
what I should do to customize the file name of the recording.
I guess some changes to the dialplan is required ?
try something like:
Monitor(wav,${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}-${EXTEN})
--
Edwin Lam edwin@officegeneral.com
Systems Engineer, OfficeWyze, Inc.
Ph: +1 415 439 4988 Fax: +1 415 283 3370
http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20
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Re: [asterisk-users] Realtime and priority labels
On 4/24/11 1:21 PM, Bruce Ferrell wrote:
In the following example
exten = _1NXXNXX,1,Set(GROUP(outbound)=myprovider)
exten = _1NXXNXX,n,Set(COUNT=${GROUP_COUNT(myprovider@outbound)})
exten = _1NXXNXX,n,NoOp(There are ${COUNT} calls for myprovider)
exten = _1NXXNXX,n,GotoIf($[ ${COUNT} 2 ]?denied : continue)
exten = _1NXXNXX,n(denied),NoOp(There are too many calls up)
exten = _1NXXNXX,n,Hangup()
exten = _1NXXNXX,n(continue),GoSub(callmyprovider,${EXTEN},1)
instead of sequentially numbering the priorities, the n construct is used. I
find that when I attempt this in the realtime extensions table only, the first
priority step is recognized. If I sequentially number the priorities and add a
label, the step is no longer recognized.
Is this behavior by design or an error?
i think it's probably by design. unlike reading from a
text file, database rely on column values for sorting.
i don't think having 'n' as the priority will sort the way
you want.
--
Edwin Lam edwin@officegeneral.com
Systems Engineer, OfficeWyze, Inc.
Ph: +1 415 439 4988 Fax: +1 415 283 3370
http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20
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Re: [asterisk-users] DTMF not being sent ( RFC2833 )
i think i have similar problem after upgraded from 1.4.x to 1.6.2.17. (originally upgraded to 1.8.3.2 unfortunately there were other more pressing problems that forced me to downgraded it to 1.6.2.17) i have a wanpipe device with 2 channels uses PRI signalling to PSTN the other 2 uses FXO signalling (connect to Rhino FXS channel bank). the PRI part works fine but the FXO channels are having DTMF digits skipped. i'm still trying to find out what's wrong with it. On 4/23/11 8:48 AM, David wrote: Hello, I installed Asterisk 1.6.2.17.3 ( latest as of yesterday ) and had multiple problems with DTMF. I have two machines, we'll call them asterisk and asterisk-pri. Asterisk does IVR and asterisk-pri has a PRI card in it and connects to the PSTN. The two servers communicate via SIP with RFC2833. I setup logger.conf on both machines to display DTMF to the console. Both are built from source. Asterisk : spandsp, dahdi, asterisk. Asterisk-pri : spandsp, libpri, dahdi, asterisk wanpipe I eliminated AGI, hard phones, network et al by setting up this extension : exten = 22,1,Dial(SIP/114186939...@pri1.omnity.net,30,D(132412983 mailto:SIP/114186939...@pri1.omnity.net,30,D(132412983#)) in default. The only other non default setting is in sip.conf I added a outboundproxy ( which does NOT do RTP, only SIP ). I called asterisk from my hard phone ( gxp2000 ) by dialing 22. I see the console DTMF messages indicating the DTMF was sent or received. ( I forgot to keep this output ). I than watch the console DTMF output on asterisk-pri and it showed about half the DTMFs. The pager that was called showed the DTMFs that appeared on the asterisk-pri console. So somewhere between the two machines, the DTMFs have disappeared. So I ran TCPDump on asterisk and saw that close to half of the DTMF events were never sent. tcpdump -i eth0 -n -s 0 dst asterisk-pri-ip -vvv -w ~/dtmf.pcap I imported the file into wireshark on my local machine and confirmed that the dump almost matches what I saw on asterisk-pri. So, problem 1 : Asterisk is not sending all the DTMFs to asterisk-pri. I compared the packet scan to what I saw on asterisk-pri and noticed that between 1 and 3 dtmfs were missing. Problem 2 : Asterisk-pri loses some received DTMFs. I also noticed that some of the DTMFs coming out of asterisk had the wrong Event Duration. I had one DTMF with a duration of about 58000 ( I believe that's 58 seconds ) but I only pressed the button for like 1/3 of a second. What I do not understand is that I in my final test last night was using asterisk 1.6 current with centos ( os that asterisk is developed on from my understanding ) with all default settings ( excluding logger.conf, dialplan and outboundproxy ) and I am having problems with the DTMF. Both servers were installed with CentOS 5.5 and were updated last night, after which I reinstalled asterisk. This did not resolve the issue. I am at wit's end and do not know where to go from here. I would really appreciate it if someone could give me some pointers on where to go next, what additionnal debugging steps I should perform. I would also really appreciate if someone could propose a solution. Please help! David Never give up, never surrender -- Edwin Lam edwin@officegeneral.com Systems Engineer, OfficeWyze, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8.3
On 4/5/11 6:10 PM, Bryant Zimmerman wrote: I have deployed several 1.8.3.2 systems as upgrades of customers systems and now I am seeing random crashes. For some reason the builds lock up and stop taking sip connections. Existing calls stay on but when the user hangs up no new calls or reg attempts work. In most cases a core restart now cleans things up. Some times I have to kill the asterisk process. The stability of 1.8.2 was poor but it is worse with 1.8.3.2 any ideas of how I can approach solving this. We've upgraded our system over the weekend from 1.4.35 to 1.8.3.2 For the past couple of days, we had several random hangs(most of the time core stop now didn't work, I had to kill -9 the process) Also the PRI behavior seems to be slightly different, we can't hear any early media sounds on 800 numbers that goes through ATT. I finally downgraded it back to 1.6.2.17, now everything work. -- Edwin Lam edwin@officegeneral.com Systems Engineer, OfficeWyze, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8.3
-6-0b45 *** i used the same SIP station to dial the same 800 number on both versions (1.8.3.2 1.6.2.17). the output are pretty much identical except on 1.8.3.2, after the PROGRESS with cause code 127... message. i would hear nothing until the other side timed out hang up, whereas on 1.6.2.17. i got the DAHDI/... is making progress passing it to SIP... message and can hear the early media from the other side. For Now 1.8.3..2 is very bad. agreed... -- Edwin Lam edwin@officegeneral.com Systems Engineer, OfficeWyze, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connecting Asterisk to Siemens Hipath 3750
On 3/10/11 6:43 AM, Bobola Oke wrote: The telco has a DB9 terminated interface straight to the PBX and I cannot make sense out of the interface for the PBX. What kind of interface is this? How do I connect the RJ48 of the PRI cards to make this whole setting work. searching through this list's archive and found this: http://lists.digium.com/pipermail/asterisk-users/2006-December/174258.html -- Edwin Lam edwin@officegeneral.com Systems Engineer, OfficeWyze, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fail-over server
On 2/9/11 6:55 AM, Vieri wrote: I'd like to do that without Realtime (or with Realtime+FreePBX) or with any other means that doesn't require more than 2 servers (2 asterisk boxes)? we use drbd nfs cluster to store asterisk's ASTDB voice mail files but that would involve installing 2 extra servers for such purpose. however you can look into csync2 to sync all asterisk files between the 2 asterisk servers if you don't want extra hardware. -- Edwin Lam edwin@officegeneral.com Systems Engineer, OfficeWyze, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SILK codec
hi folks. i've been experimenting with SILK codec and meet with some success on incorporating it in pjsip (an open source sip client). now i'm trying to do the same thing on Asterisk. any documentations, pointers, etc i should look into? any help is appreciated. -- Edwin Lam edwin@officegeneral.com Systems Engineer, OfficeWyze, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Configuring Softphone
On 12/7/10 10:20 PM, Jeroen Eeuwes wrote: host = dynamic defaultip = dynamic The defaultip line says to Asterisk that this phone is found at the IP-address associated with the hostname dynamic. I'm pretty sure that if you try a command like ping dynamic on your command line it will not return the IP address of your phone. actually host = dynamic is correct if the client doesn't have a fixed ip address. however defaultip should set at a fixed ip address if the client is not registered from anywhere. normally i'd just omit the defaultip parameter. also is your softphone and/or asterisk box behind NAT? -- Edwin Lam edwin@officegeneral.com Systems Engineer, OfficeWyze, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] changing sip port
On 11/11/10 2:25 PM, Baha @ SH wrote: How can I run the sip service on asterisk on another port beside 5080? I mean asterisk will still take sip requests on port:5080 and another custom port, lets say port:6080 you can only configure 1 listening port in sip.conf. however you can use port forwarding on OS level to achieve that. iptables -t nat -I PREROUTING -i you incoming interface -d your IP addr -p udp -m udp --dport 6080 -j DNAT --to-destination your ip addr:5060 p.s. the default SIP port is 5060. -- Edwin Lam edwin@officegeneral.com Systems Engineer, OfficeWyze, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] take input and store in variable
On 10/4/10 12:27 PM, Tom Lohmuller wrote:
I am using a context to change values in a DB. Currently in my context, I
am passing it to
exten = s,1,WaitExten(7) ; 7 seconds to input
exten = s,n,Set(NEW_VAR=${EXTEN}) ;Here is my problem. This is the only
way I know how to 'grab' user input, which was normally from ${EXTEN} but
I realize this won't work for extension 's'..
The short google search I did didn't turn up anything concrete.
try:
exten = s,1,WaitExten(7)
exten = _X!,1,Set(NEW_VAR=${EXTEN})
exten = _X!,n,do other things...
.
.
exten = t,1,Hangup() ;hang up if no input for 7 sec.
--
Edwin Lam edwin@officegeneral.com
Systems Engineer, OfficeWyze, Inc.
Ph: +1 415 439 4988 Fax: +1 415 283 3370
http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] more condition check for gotoif
On 10/3/10 11:20 AM, Daniel Knoll wrote: Hello, is it possible to check more than one condition for GOTOIF in the dialplan? yes. check out asterisk expressions on wiki pages http://www.voip-info.org/wiki/view/Asterisk+Expressions -- Edwin Lam edwin@officegeneral.com Systems Engineer, OfficeWyze, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DAHDI fxstest?
On 9/9/10 1:40 PM, Tim Nelson wrote: During some recent testing and debugging, I wanted to use the 'fxstest' application. However, I found it hasn't been built when doing the standard 'make, make install' shtick with dahdi-linux-complete-2.3.0.1+2.3.0... Can anyone tell me how to build fxstest? cd to tools directory then enter the command make menuselect select fxstest, save exit, then execute make. -- Edwin Lam edwin@officegeneral.com Systems Engineer, OfficeWyze, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] early media issue from phone co.
hi folks. i have the following puzzle: when i call certain cell phone# using a regular phone POTS. the called cell phone co. usually return a message such as phone travel out of range or phone is busy etc. if the phone is unreachable. now when i have the following setup: sip phone - asterisk - PRI - phone co. i call the same cell# and if it's unavailable. the PRI return cause code 31 and hangup, asterisk will then send a SIP BYE to the sip phone and the channel will simply hangup. how do i get the message on the sip phone? -- Edwin Lam edwin@officegeneral.com Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fax spandsp
Steve Underwood wrote: Crashes of this kind are not uncommon, but the causes are: - Multiple versions of libtiff installed in different directories checked that. got only single version. - Multiple versions of spandsp installed in different directories i've got this: Hydra:/usr/local/lib# ls -l libspandsp* -rw-r--r-- 1 root staff 2098532 2010-03-07 17:05 libspandsp.a -rwxr-xr-x 1 root staff 865 2010-03-07 17:05 libspandsp.la lrwxrwxrwx 1 root staff 19 2010-03-07 17:05 libspandsp.so - libspandsp.so.2.0.0 lrwxrwxrwx 1 root staff 19 2006-08-02 08:35 libspandsp.so.0 - libspandsp.so.0.0.1 -rwxr-xr-x 1 root staff 976884 2006-08-02 08:35 libspandsp.so.0.0.1 lrwxrwxrwx 1 root staff 19 2010-03-07 17:05 libspandsp.so.2 - libspandsp.so.2.0.0 -rwxr-xr-x 1 root staff 1597979 2010-03-07 17:05 libspandsp.so.2.0.0 but whereis command only show 1 version. Hydra:/usr/local/lib# whereis libspandsp libspandsp: /usr/local/lib/libspandsp.a /usr/local/lib/libspandsp.la /usr/local/lib/libspandsp.so should i delete the .0.0.1 just in case? - Asterisk was built against a spandsp installed in a directory that is not in the library search path, while another version of spandsp is in a directory that is in the library search path. so, at run time the wrong library is picked up. Many machines will happily build and install a library to /usr/local, and then successfully like applications against it, even though /usr/local is not in the library search path. Dumb, but true. The installation information page for spandsp, at http://www.soft-switch.org/installing-spandsp.html , warns about these issues. -- Edwin Lam edwin@officegeneral.com Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fax spandsp
i gave up on ReceiveFAX and uses iaxmodem/hylafax instead. Tommy Botten Jensen wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA512 Edwin Lam skrev: Klaus Darilion wrote: The backtrace is not useable. Try to rebuild Asterisk with the Don't Optimize Option (make menuconfig and the the build options) did that. no effect. i've got exactly the same result. Edwin Lam wrote: Philip A. Prindeville wrote: On 03/08/2010 04:31 PM, Edwin Lam wrote: hi folks. i recently upgraded asterisk to 1.6.1.17(from 1.2) and i'm having problems with fax. after receiving fax with the ReceiveFAX app. everything seems ok. the .tiff file was there, phone line seems to hang up. then asterisk will crash. any ideas? also i looked in the log file. this is what before it crashed: [Mar 8 12:12:12] WARNING[30115] app_fax.c: WARNING T.30 ECM carrier not found [Mar 8 12:12:12] WARNING[30115] app_fax.c: WARNING T.30 ECM carrier not found [Mar 8 12:12:12] WARNING[30115] app_fax.c: WARNING T.30 ECM carrier not found [Mar 8 12:12:12] WARNING[30115] app_fax.c: WARNING T.30 ECM carrier not found [Mar 8 12:12:30] VERBOSE[30115] pbx.c: [Mar 8 12:12:30] -- Auto fallthrough, channel 'DAHDI/8-1' status is 'UNKNOWN' [Mar 8 12:12:30] VERBOSE[30115] pbx.c: [Mar 8 12:12:30] -- Executing [...@detectfax:1] GotoIf(DAHDI/8-1, 1?200) in new stack [Mar 8 12:12:30] VERBOSE[30115] pbx.c: [Mar 8 12:12:30] -- Goto (detectfax,h,200) [Mar 8 12:12:30] VERBOSE[30115] pbx.c: [Mar 8 12:12:30] -- Executing [...@detectfax:200] System(DAHDI/8-1, /usr/local/bin/mailfax /var/spool/asterisk/fax/4502-1268079069.417.tif x...@.com ) in new stack [Mar 8 12:12:31] VERBOSE[30115] chan_dahdi.c: [Mar 8 12:12:31] -- Hungup 'DAHDI/8-1' asterisk: 1.6.1.17 spandsp: 0.0.6pre17 What happens when you turn off autofallthrough? exactly same thing except instead of the Auto fallthrough line the following came up: pbx.c:3928 __ast_pbx_run: Don't know what to do with 'DAHDI/5-1' and also here's the backtracce (i'm using Debian lenny) *** glibc detected *** /usr/sbin/asterisk: double free or corruption (!prev): 0x082528b8 *** === Backtrace: = /lib/i686/cmov/libc.so.6[0xb7d66624] /lib/i686/cmov/libc.so.6(cfree+0x96)[0xb7d68826] /usr/sbin/asterisk[0x80d2e89] /lib/i686/cmov/libpthread.so.0[0xb7ce156a] /lib/i686/cmov/libc.so.6(clone+0x5e)[0xb7dd86de] I am running Asterisk 1.6.2.1 and 1.6.2.2 with Spandsp 0.0.6-pre17 with the exact same error messages. But my asterisk does not have any trouble apart from the messages themselves. I did however run into this earlier, and I believe it was fixed for the 1.6.2.x series. At least it worked for me. Best regards, Tommy Botten Jensen -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iEYEAREKAAYFAkuXYlwACgkQ573V05EH/pZpZQCeO8FPGqAJ4cRDlnyZOERbgNoj 0TEAmgOiY0byfIy3SIM5GR9gDrG+LZEY =oN/L -END PGP SIGNATURE- -- Edwin Lam edwin@officegeneral.com Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fax spandsp
Klaus Darilion wrote: The backtrace is not useable. Try to rebuild Asterisk with the Don't Optimize Option (make menuconfig and the the build options) did that. no effect. i've got exactly the same result. Edwin Lam wrote: Philip A. Prindeville wrote: On 03/08/2010 04:31 PM, Edwin Lam wrote: hi folks. i recently upgraded asterisk to 1.6.1.17(from 1.2) and i'm having problems with fax. after receiving fax with the ReceiveFAX app. everything seems ok. the .tiff file was there, phone line seems to hang up. then asterisk will crash. any ideas? also i looked in the log file. this is what before it crashed: [Mar 8 12:12:12] WARNING[30115] app_fax.c: WARNING T.30 ECM carrier not found [Mar 8 12:12:12] WARNING[30115] app_fax.c: WARNING T.30 ECM carrier not found [Mar 8 12:12:12] WARNING[30115] app_fax.c: WARNING T.30 ECM carrier not found [Mar 8 12:12:12] WARNING[30115] app_fax.c: WARNING T.30 ECM carrier not found [Mar 8 12:12:30] VERBOSE[30115] pbx.c: [Mar 8 12:12:30] -- Auto fallthrough, channel 'DAHDI/8-1' status is 'UNKNOWN' [Mar 8 12:12:30] VERBOSE[30115] pbx.c: [Mar 8 12:12:30] -- Executing [...@detectfax:1] GotoIf(DAHDI/8-1, 1?200) in new stack [Mar 8 12:12:30] VERBOSE[30115] pbx.c: [Mar 8 12:12:30] -- Goto (detectfax,h,200) [Mar 8 12:12:30] VERBOSE[30115] pbx.c: [Mar 8 12:12:30] -- Executing [...@detectfax:200] System(DAHDI/8-1, /usr/local/bin/mailfax /var/spool/asterisk/fax/4502-1268079069.417.tif x...@.com ) in new stack [Mar 8 12:12:31] VERBOSE[30115] chan_dahdi.c: [Mar 8 12:12:31] -- Hungup 'DAHDI/8-1' asterisk: 1.6.1.17 spandsp: 0.0.6pre17 What happens when you turn off autofallthrough? exactly same thing except instead of the Auto fallthrough line the following came up: pbx.c:3928 __ast_pbx_run: Don't know what to do with 'DAHDI/5-1' and also here's the backtracce (i'm using Debian lenny) *** glibc detected *** /usr/sbin/asterisk: double free or corruption (!prev): 0x082528b8 *** === Backtrace: = /lib/i686/cmov/libc.so.6[0xb7d66624] /lib/i686/cmov/libc.so.6(cfree+0x96)[0xb7d68826] /usr/sbin/asterisk[0x80d2e89] /lib/i686/cmov/libpthread.so.0[0xb7ce156a] /lib/i686/cmov/libc.so.6(clone+0x5e)[0xb7dd86de] -- __ Edwin Lam edwin@officegeneral.com __ __ Systems Engineer, Office General, Inc. __ Ph: +1 415 439 4988 Fax: +1 415 283 3370 __ __ http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0xEF11A895 __ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] fax spandsp
hi folks. i recently upgraded asterisk to 1.6.1.17(from 1.2) and i'm having problems with fax. after receiving fax with the ReceiveFAX app. everything seems ok. the .tiff file was there, phone line seems to hang up. then asterisk will crash. any ideas? also i looked in the log file. this is what before it crashed: [Mar 8 12:12:12] WARNING[30115] app_fax.c: WARNING T.30 ECM carrier not found [Mar 8 12:12:12] WARNING[30115] app_fax.c: WARNING T.30 ECM carrier not found [Mar 8 12:12:12] WARNING[30115] app_fax.c: WARNING T.30 ECM carrier not found [Mar 8 12:12:12] WARNING[30115] app_fax.c: WARNING T.30 ECM carrier not found [Mar 8 12:12:30] VERBOSE[30115] pbx.c: [Mar 8 12:12:30] -- Auto fallthrough, channel 'DAHDI/8-1' status is 'UNKNOWN' [Mar 8 12:12:30] VERBOSE[30115] pbx.c: [Mar 8 12:12:30] -- Executing [...@detectfax:1] GotoIf(DAHDI/8-1, 1?200) in new stack [Mar 8 12:12:30] VERBOSE[30115] pbx.c: [Mar 8 12:12:30] -- Goto (detectfax,h,200) [Mar 8 12:12:30] VERBOSE[30115] pbx.c: [Mar 8 12:12:30] -- Executing [...@detectfax:200] System(DAHDI/8-1, /usr/local/bin/mailfax /var/spool/asterisk/fax/4502-1268079069.417.tif x...@.com ) in new stack [Mar 8 12:12:31] VERBOSE[30115] chan_dahdi.c: [Mar 8 12:12:31] -- Hungup 'DAHDI/8-1' asterisk: 1.6.1.17 spandsp: 0.0.6pre17 -- Edwin Lam edwin@officegeneral.com Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fax spandsp
Philip A. Prindeville wrote: On 03/08/2010 04:31 PM, Edwin Lam wrote: hi folks. i recently upgraded asterisk to 1.6.1.17(from 1.2) and i'm having problems with fax. after receiving fax with the ReceiveFAX app. everything seems ok. the .tiff file was there, phone line seems to hang up. then asterisk will crash. any ideas? also i looked in the log file. this is what before it crashed: [Mar 8 12:12:12] WARNING[30115] app_fax.c: WARNING T.30 ECM carrier not found [Mar 8 12:12:12] WARNING[30115] app_fax.c: WARNING T.30 ECM carrier not found [Mar 8 12:12:12] WARNING[30115] app_fax.c: WARNING T.30 ECM carrier not found [Mar 8 12:12:12] WARNING[30115] app_fax.c: WARNING T.30 ECM carrier not found [Mar 8 12:12:30] VERBOSE[30115] pbx.c: [Mar 8 12:12:30] -- Auto fallthrough, channel 'DAHDI/8-1' status is 'UNKNOWN' [Mar 8 12:12:30] VERBOSE[30115] pbx.c: [Mar 8 12:12:30] -- Executing [...@detectfax:1] GotoIf(DAHDI/8-1, 1?200) in new stack [Mar 8 12:12:30] VERBOSE[30115] pbx.c: [Mar 8 12:12:30] -- Goto (detectfax,h,200) [Mar 8 12:12:30] VERBOSE[30115] pbx.c: [Mar 8 12:12:30] -- Executing [...@detectfax:200] System(DAHDI/8-1, /usr/local/bin/mailfax /var/spool/asterisk/fax/4502-1268079069.417.tif x...@.com ) in new stack [Mar 8 12:12:31] VERBOSE[30115] chan_dahdi.c: [Mar 8 12:12:31] -- Hungup 'DAHDI/8-1' asterisk: 1.6.1.17 spandsp: 0.0.6pre17 What happens when you turn off autofallthrough? exactly same thing except instead of the Auto fallthrough line the following came up: pbx.c:3928 __ast_pbx_run: Don't know what to do with 'DAHDI/5-1' and also here's the backtracce (i'm using Debian lenny) *** glibc detected *** /usr/sbin/asterisk: double free or corruption (!prev): 0x082528b8 *** === Backtrace: = /lib/i686/cmov/libc.so.6[0xb7d66624] /lib/i686/cmov/libc.so.6(cfree+0x96)[0xb7d68826] /usr/sbin/asterisk[0x80d2e89] /lib/i686/cmov/libpthread.so.0[0xb7ce156a] /lib/i686/cmov/libc.so.6(clone+0x5e)[0xb7dd86de] -- Edwin Lam edwin@officegeneral.com Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] splitting sip.conf to two files
Joseph wrote: Is it possible to split sip.conf into two files (sip1.conf sip2.conf)? I have an Audiocodes gateway with two FXO ports, and (according to info I received, and it appears to be correct) Asterisk find the peers based on their IP and not on their IP+PORT. Thus, Audiocodes with two FXO ports registered on the same devices (= one single IP with different SIP ports), the last entry into my sip.conf file is taken into consideration = all calls are sent to the context of that last extension. So I can only use one context for incoming calls. If I split the sip.conf into two files will it make any difference. there might be an include directive in sip.conf (i can't confirm) however Asterisk will see it as one big sip.conf so it will do absolutely nothing for you in this situation. what you can do is setup automatic dial to different extensions on the 2 ports on audiocodes. -- Edwin Lam edwin@officegeneral.com Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] server unresponsive
hi folks. we've experienced some weird problems lately. we have about 600 SIP phone on a single system running *1.4.26.2 for about a month. recently there was massive UNREACHABLE messages like this one showed up: chan_sip.c: Peer '2699' is now UNREACHABLE! Last qualify: 1252 then they all became reachable again in a few seconds. sometimes it last for couple minutes. but sometimes it last for hours, when that happens. the system will get very slow and eventually error like this will start showing: channel.c: Exceptionally long voice queue length queuing to IAX2/hostpbx2-12619 after a while the whole system will become unresponsive until i kill the asterisk process. i've checked our network switches/routers and connections. they all work fine without any packet lost. any suggestions? -- Edwin Lam edwin@officegeneral.com Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] server unresponsive
Paul Scott wrote:
How is your network structured?
we have a central location where the * server is located. and 4 remote
locations connected via point to point lines.
Can you show me a sample entry from your sip.conf?
we use realtime sip w/ mysql tables. a typical entry would looks
like this:
[1234]
context=default10
type=friend
secret=xyz
qualify=yes
host=dynamic
canreinvite=no
dtmfmode=rfc2833
disallow=all
allow=g729
allow=alaw
allow=ulaw
I was having this problem. But as far as I could tell there wasn't
one. From a network stand point all phones were reachable asterisk
was just reporting that it was Unreachable and it wasn't sending the
calls. I switched to qualify=no and wrote a small agi to catch $
{HANGUPCAUSE} and log it to a file. If it records a bunch of
chanunavail messages you still have a problem.
If you don't want to turn qualify off you could play with the qualify
times. I did a bunch of this before I just gave up.
I'm sure there is a better or proper way of handling this. I'm
interested to hear it.
Paul
On Nov 20, 2009, at 3:41 PM, Edwin Lam wrote:
hi folks.
we've experienced some weird problems lately. we have about 600
SIP phone on a single system running *1.4.26.2 for about a month.
recently there was massive UNREACHABLE messages like this one
showed up:
chan_sip.c: Peer '2699' is now UNREACHABLE! Last qualify: 1252
then they all became reachable again in a few seconds. sometimes
it last for couple minutes. but sometimes it last for hours, when
that happens. the system will get very slow and eventually error
like this will start showing:
channel.c: Exceptionally long voice queue length queuing to IAX2/
hostpbx2-12619
after a while the whole system will become unresponsive
until i kill the asterisk process.
i've checked our network switches/routers and connections.
they all work fine without any packet lost.
any suggestions?
--
Edwin Lam edwin@officegeneral.com
Systems Engineer, Office General, Inc.
Ph: +1 415 439 4988 Fax: +1 415 283 3370
http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20
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[asterisk-users] transcoder card
hi folks. i have several remote sites with total of 200 sip phones connect to our Asterisk server. i want to minimize bandwidth usage and thinking about getting a Digium TC400B transcoder card. what are your experience with it? how's the quality? also if there are 120 active channels in used. will the 121 person able to make calls? will it support more channels if i put 2 cards in the system? thanks. -- Edwin Lam edwin@officegeneral.com Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Time of Day Routing
Dáibhéad Antoine O'Reilligh wrote:
I have a logic question that is confusing me.
ifTime(00:00-12:00|*|*|*) {
Playback(welcome-morning);
} else {
ifTime(12:00-18:00|*|*|*) {
Playback(welcome-afternoon);
} else {
Playback(welcome-evening);
}
}
Does that cover the entire day? The question arose because of the time
as follows:
ifTime(00:00-12:00|*|*|*) { so midnight to noon fine--but should
it be 11:59 or 12:00
and ifTime(12:00-18:00|*|*|*) { -- should it be from 12:01
However doing that I assume will miss a minute.
according to the wiki pages:
minute = a number, 0 to 59, inclusive
so. look back at your logic, your welcome-morning message will
be played before 12:01. and your welcome-afternoon message will
be played on after 12:01.
--
Edwin Lam edwin@officegeneral.com
Systems Engineer, Office General, Inc.
Ph: +1 415 439 4988 Fax: +1 415 283 3370
http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20
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Re: [asterisk-users] Here is Step by Step Example of Asterisk PBX System Install and configuration
Tilghman Lesher wrote: On Friday 17 April 2009 16:33:21 Steve Edwards wrote: I like to use vi because it is a common editor should be I suffer with vi because I never learned to use a real editor like emacs --- ducking quickly :) Emacs is a nice operating system, but it lacks a decent editor. :-P forget about all these editors.. TECO is the Sh*t. -- Edwin Lam edwin@officegeneral.com Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IMAP voicemail storage.
i've been playing with 1.6 voicemail w/ IMAP storage. it seems to work fine. however once IMAP storage is enabled. everyone VM will use IMAP. is there a way to configure some users use IMAP and other users use traditional file base storage? -- Edwin Lam edwin@officegeneral.com Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk/E1
Khaled Chehab wrote: Dears My hitch is that no alerts bombing out .all what is bombing out Unable to open pid file '/var/run/asterisk/asterisk.pid': No such file or directory check to see if the directory /var/run/asterisk exist and the permissions are correct if you run asterisk as non-root. -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] a little regex help needed
sean darcy wrote:
OK. So I changed the * to .. , like so:
exten =s,n,Set(CALLERID(name)=${IF($[0${CALLERID(num)} =
0${REGEX(21245711..)} ] ? Office:${CALLERID(name)} )})
which I would expect to mean 021245711 followed by two other characters.
It still matches a blank callerid(num).
try:
exten =s,n,Set(CALLERID(name)=${IF($[ ${REGEX(021245711..
0${CALLERID(num)})} = 1] ? Office:${CALLERID(name)})})
--
Edwin Lam [EMAIL PROTECTED]
Systems Engineer, Office General, Inc.
Ph: +1 415 439 4988 Fax: +1 415 283 3370
http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20
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Re: [asterisk-users] Newbie in China: Red alaram in Zaptel for E1
Lee, John (Sydney) wrote: i've installed several Asterisk systems in Shanghai Beijing. Thanks Edwin. The remote site is in Shanghai and NETCOM is the telco. Do you know if their E1 line is MFC/R2 or EuroISDN? i'm not sure if they provide MFC/R2. but we always ordered PRI from them. as far as switch type. seems like nobody in CNC can give us a definite answer, but we have success using EuroISDN swicth type. red alarm usually means there's no clocking signal. check all your cables (crossover vs straight through) As far as the cable goes, this is a bit complicated. The way it works is the telco delivers a fibre optic cable to the floor and the fibre terminates on a fibre optic multiplexer. Then the multiplexer is connected to a Fast Ethernet to E1 converter which has a RJ45 port. We then connect this RJ45 port to the TE412P port. Anyway what you said is still a good point - I will try replacing the straight through cable with a crossover and give it a go. if the cable's good. call phone company and complain. in my experience 9 out of 10 time we have to call phone company and complain. How should we complain? Are there any technical details we need to show them? It is a different country though. if after you tried both straight through crossover cables and it still give you RED alarm. just tell them you can't get any clocking signal. they'll probably send someone on site and test the line. p.s. note that T1/E1 crossover cable pin out is not the same as ethernet crossover cable. -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie in China: Red alaram in Zaptel for E1
Lee, John (Sydney) wrote: This time, I am trying to remotely install Asterisk in China. I was told that an E1 line has been installed and so I plug it into port 1 of a TE412P. i've installed several Asterisk systems in Shanghai Beijing. On the box, first of all, I just installed Zaptel 1.4.10.1. # service zaptel restart Unloading zaptel hardware drivers:ERROR: Module zaptel is in use . Loading zaptel framework: [ OK ] Waiting for zap to come online...OK Loading zaptel hardware modules: tor2. wct4xxp. wcte12xp. wct1xxp. wcte11xp. wctdm24xxp. wcfxo. wctdm. wcusb. Running ztcfg: [ OK ] you might want to comment out all other modules in /etc/default/zaptel except for wct4xxp (if that's the only zaptel card you have). # vi zaptel.conf [...] span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 this looks right. however crc4 is optional. you have to check with the phone company. sometime they do require it other time they don't. it's not very consistent. *** However, I received a red alarm in zttool and the LED on the TE412P card is also red. *** I have made sure that the jumper is closed for port 1 on the TE412P card and so it could not be the jumper problem. red alarm usually means there's no clocking signal. check all your cables (crossover vs straight through) if the cable's good. call phone company and complain. in my experience 9 out of 10 time we have to call phone company and complain. ### Because this is the first time I install Asterisk in China and I was wondering if their E1 is different from the Euro E1. ### However, I went into dmesg and I discovered the following. ### Could it really be a zaptel bug? I saw on a similar few on the digium bug list but I cannot be 100% sure. Any thoughts? About to enter spanconfig! Done with spanconfig! Registered tone zone 33 (China) About to enter startup! TE4XXP: Span 1 configured for CCS/HDB3/CRC4 timing source auto card 0! wct4xxp: Setting yellow alarm on span 1 timing source auto card 0! VPM400: Not Present VPM450: echo cancellation for 128 channels BUG: soft lockup - CPU#2 stuck for 16s! [ztcfg:4681] Pid: 4681, comm:ztcfg EIP: 0060:[f8cba1df] CPU: 2 EIP is at init_vpm450m+0x32d/0x34a [wct4xxp] EFLAGS: 0286Tainted: G (2.6.18-92.1.6.el5 #1) EAX: EBX: f76ae8f0 ECX: 0019 EDX: ESI: f7997400 EDI: 0286 EBP: f76838c0 DS: 007b ES: 007b CR0: 8005003b CR2: b7f01007 CR3: 37bc1000 CR4: 06d0 [f8ca1b11] t4_vpm450_init+0x18ce/0x198c [wct4xxp] [f8ca5ee4] t4_startup+0x4315/0x43c7 [wct4xxp] [c042609c] release_console_sem+0x17e/0x1b8 [c046d53a] cache_alloc_refill+0x14b/0x450 [f8956f61] zt_ioctl+0x273/0x144f [zaptel] [c04d7d45] generic_make_request+0x248/0x258 [c045ae3c] __do_page_cache_readahead+0x69/0x1c6 [c0484a5b] __d_lookup+0x98/0xdb [c047c110] do_lookup+0x53/0x166 [c047e7e4] do_path_lookup+0x20e/0x25e [c047c389] permission+0xa2/0xb5 [c04e2d06] kobject_get+0xf/0x13 [c046f7fa] __dentry_open+0xea/0x1ab [c046f91f] nameidata_to_filp+0x19/0x28 [c046f959] do_filp_open+0x2b/0x31 [c048029b] do_ioctl+0x47/0x5d [c04804fb] vfs_ioctl+0x24a/0x25c [c0471bbe] __fput+0x13f/0x167 [c0480555] sys_ioctl+0x48/0x5f [c0404eff] syscall_call+0x7/0xb === VPM450: hardware DTMF disabled. VPM450: Present and operational servicing 4 span(s) Completed startup! i've seen that before. (forgot which version of zaptel). it went away after i upgraded it. -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] iax2_trunk_queue: Maximum trunk data space exceeded
hi folks. one of the servers i setup recently start exhibiting iax2_trunk_queue: Maximum trunk data space exceeded errors. there was only 1 call going on at the time. usually i have to reload chan_iax2.so or restart Asterisk. but the errors came back within a few minutes. i did a google search on that error. but nothing useful came up. i'm kinda stuck. anyone have any ideas? p.s. hardware: HP proliant BL460c server OS: Linux 2.6.18 (Debian Etch AMD64 customize kernel) Asterisk: 1.4.18 i have several setup with the same version of everything except the hardware slightly different. they never have this kind of problems. -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Data Calls
Brent Davidson wrote: I'm using the Rhino R4FXO cards to answer incoming voice calls and just had this idea last night... Is there any type of module for asterisk that will divert a call to a softmodem? For example, I call in on the voice lines, get the main menu, instead of dialing an extension to get a person I dial an extension and am given modem tones. ( I basically want to have a way to dial in to my offices without having to have dedicated data lines and RAS servers. I know I could probably make this work by sticking a modem in the box and using an FXS port to route to the call to the modem, but that is messy, in my opinion, and I would like to avoid that if possible. Is this at all possible? (I don't want to route the data audio over IP. I know that would be largely impossible.) will this work for you? http://sourceforge.net/projects/iaxmodem -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Out-Going Calleriid
Tim Guy wrote:
Installing a new box onto UK NTL (Virgin Media)
During testing phase the callerid worked, now it doesn't.
Can someone confirm that my syntax is right before I start ripping the
configs to bits
exten = _9.,1,Set(CALLERID(number)=01926xx)
exten = _9.,2,Dial(ZAP/1/${EXTEN:1})
Ive tried all permutations of the CALLERID (ie CALLERID(NAME) and
CALLERID(NUMBER) but it just wont work anymore.
Zapata has the following relevant settings
usecallerid=yes
hidecallerid=no
callwaiting=yes
you may need to add
usecallingpres=yes
in zapata.conf
and also add
exten = _9.,n,SetCallerPres(allow)
before the Dial command in extensions.conf.
--
Edwin Lam [EMAIL PROTECTED]
Systems Engineer, Office General, Inc.
Ph: +1 415 439 4988 Fax: +1 415 283 3370
http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20
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Re: [asterisk-users] iaxmodem + hylafax w/ DID routing
Doug Lytle wrote: Edwin Lam wrote: in FaxDispatch: FILETYPE=pdf case $CALLID4 in 1000) [EMAIL PROTECTED] 1001) [EMAIL PROTECTED] *) [EMAIL PROTECTED] esac This is also incomplete, actually there's no problems with the above script. it's just $CALLID4 always return blank, causing the email send to the default $SENDTO address. One of my entries with archiving of the PDF and TIF: case $CALLID4 in '5051') # ## Bankers Life/Conseco (Rose Parker) (Previously Louise Taylor)# # FILETYPE=pdf; [EMAIL PROTECTED]; /usr/local/bin/tiff2pdf $FULLPATH -p letter -o faxdata/$CALLID4/pdf/$FILENAME.pdf cp $FULLPATH /var/spool/hylafax/faxdata/$CALLID4/tif/ ;; Doug -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] iaxmodem + hylafax w/ DID routing
hi folks.
i'm experimenting with iaxmodem + hylafax using DID to determine
where to send the fax to it's final destination. however i have
difficulties passing the DID information from iaxmodem to
hylafax.
in extensions.conf:
exten = _,1,Dial(IAX2/iaxmodem0/${EXTEN}|20|r)
exten = _,n,Dial(IAX2/iaxmodem1/${EXTEN}|20|r)
exten = _,n,Busy
exten = _,n,Hangup
in FaxDispatch:
FILETYPE=pdf
case $CALLID4 in
1000)
[EMAIL PROTECTED]
1001)
[EMAIL PROTECTED]
*)
[EMAIL PROTECTED]
esac
according to some documentations i've found $CALLID4
should have the DID info. but in fact it's blank.
any idea?
--
Edwin Lam [EMAIL PROTECTED]
Systems Engineer, Office General, Inc.
Ph: +1 415 439 4988 Fax: +1 415 283 3370
http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20
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Re: [asterisk-users] Need some input for Quad T1 and channel banks.
Andrew Latham wrote: Here I will say it http://xorcom.com alternatively: http://www.audiocodes.com/objects/30010_DS_MP-11X,%20MP-124D.pdf On Mon, Mar 31, 2008 at 6:01 PM, Al lists [EMAIL PROTECTED] wrote: I'm looking to install a system with 80 FXS analog phones. At this time the only cost effective solution is using a 4 port T1 card and addit 600 channel bank. Has anyone tried this solution? any good documents beside http://www.voip-info.org/wiki/index.php?page=Asterisk+hardware+channel+bank+check as far as i know, addit 600 T1 interface is not PRI (please correct me if i'm wrong) its CAS robbed bit, will that work with new Digium quad T1 like TE410P ?( I prefer to use Digium if possible) The system is connected to the Telco through SIP trunk so all we have in terms of analog is local loop, Do we need to have echo cancel in this scenario ? -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TXFax/RXFax/AGX-Addons/SpanDSP Crashing
Matt Riddell wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Tzafrir Cohen wrote: Let's be more specific here, folks: What version numbers? Asterisk, spandsp, agx-addons / rx-tx-fax? Asterisk: yesterday's 1.4 SVN SpanDSP: tried with pre 15, 16 and 18 AGX-Addons: tried with 1.4.5 and svn trunk rx/txfax: supplied by AGX Addons - although they seem to build the files and stick them into the modules directory, rather than adding to the apps directory and modifying the Makefile. i have Asterisk 1.4.18, SpanDSP 0.0.4pre16, AGX addons 1.4.5 linux kernel 2.6.18 AMD64. it (Asterisk) segfault on rxfax when i enable faxdetect in zapata.conf. since then it disabled faxdetect and use nvfaxdetect function in dialplan, it works fine afterward. also it seems to works fine using regular 32bit kernel. -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Global Variables on Reload
Rob Schall wrote: I'm running Asterisk 1.4.18 and having a problem with the clearglobalvars option. I have a NIGHT_SERVICE variable which I initially set equal to off. I then have an extension they can dial which will toggle that variable. My problem is when you enter the CLI and type reload, it resets to off again. I've tried setting the clearglobalvars=no as well as just commenting out that line, but no luck so far. Any ideas? we use MySQL db to store those global vars in our installation. i guess you can use and db to do that. -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4 w/ realtime static zapata
i've been using *1.2 w/ realtime static zapata in mysql table fine. but after i upgraded to 1.4. it seems like the zapata table doesn't load correctly. i have to go in the console and use the zap restart to get the zap channels register. is this sounds like a bug or something i'm missing when upgrading to 1.4? -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] spandsp/tx_fax/rx_fax frustrations
hi does any body know which version combination of spandsp/tx_fax/rx_fax will work with * 1.2.24? i tried different combo. they're either seg fault during runtime or won't compile. very frustrated :/ p.s. i know. hylafax/iaxmodem is far more stable. but i have specific reasons to use rx_fax. -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pulling my hair out over voicemail
John Von Essen wrote: Here are my configs: sip.conf: [general] context=default bindport=5060 bindaddr=0.0.0.0 disallow=all allow=ulaw [6000] type=friend secret=letmein host=dynamic dtmfmode=rfc2833 mailbox=6000 context=default extensions.conf: [default] exten = 1000,1,Ringing exten = 1000,2,Wait(2) exten = 1000,3,VoicemailMain Calling from phone to phone is fine, and inbound and outbound calling is fine. But when I call voicemail, I dont hear anything. When I view console in CLI I see this when attempting to dial the voicemail extension: -- Executing [EMAIL PROTECTED]:1] Ringing(SIP/6001-081d65c8, ) in new stack -- Executing [EMAIL PROTECTED]:2] Wait(SIP/6001-081d65c8, 2) in new stack -- Executing [EMAIL PROTECTED]:3] VoiceMailMain(SIP/6001-081d65c8, [EMAIL PROTECTED]) in new stack -- SIP/6001-081d65c8 Playing 'vm-login' (language 'en') [Jan 31 06:42:49] WARNING[8513]: app_voicemail.c:6281 vm_authenticate: Couldn't read username Really destroying SIP dialog '[EMAIL PROTECTED]' Method: BYE So it plays the greetings, and is working, I just cant hear it. what's your voicemail.conf looks like? also check the file permission and make sure asterisk can read it. -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cisco 79xx XML services
hi guys. i'm writing some simple applications for the cisco 7970 services button. i read the asterisk wiki and it mention there's a CMXML_App_Guide.pdf file but there's nowhere can i find a link for it. does anybody know where can i find it? regards. -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IBM x3400 w/ Digium TE220
Matthew Fredrickson wrote: This looks like a really good reason to call Digium tech support :-) It's comes free with the purchase of the card. I haven't heard of anything like this, although posting your kernel panic output would help. But it would be best to handle this through tech support. yes. i did. the problem solved by getting the latest snapshot with subversion. -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem registering Cisco 7970 phone with Asterisk 1.4 running FreePBX
John Constalgie wrote: My updated SEPMAC file for this hard phone is at http://cid-ff3ef0764138e401.skydrive.live.com/self.aspx/Public/SEP001E4A5F1270.cnf.xml try set the backup, emergency, and outbound proxies to blank under sipProxies section: sipProxies backupProxy/backupProxy backupProxyPort/backupProxyPort emergencyProxy/emergencyProxy emergencyProxyPort/emergencyProxyPort outboundProxy/outboundProxy outboundProxyPort/outboundProxyPort registerWithProxytrue/registerWithProxy /sipProxies -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem registering Cisco 7970 phone with Asterisk 1.4 running FreePBX
John Constalgie wrote: Hi Edwin, I did what you said for the SEP file ( updated SEP xml file : http://cid-ff3ef0764138e401.skydrive.live.com/self.aspx/Public/SEP001E4A5F1270.cnf.xml ) By the way, I was reading up online that I could change the qualify=yes setting to no in sip_additional.conf to make my phone unmonitored. My sip_additional.conf is now : [2001] type=friend password=2001 record_out=Adhoc record_in=Adhoc qualify=no port=5060 pickupgroup= nat=yes i believe you also need to set nat=no. on my setup (i'm using asterisk 1.2.24) it works with both qualify=yes or no. -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IBM x3400 w/ Digium TE220
hi folks. i have a Digium TE220 PCI-E 2 port T1/E1 controller installed in an IBM x3400 server. i load the wct4xxp driver seems ok. but when i execute ztcfg -vvv command. the kernel panic. i tried zaptel 1.2.21 22. they have the same result. following is my zaptel.conf: loadzone=cn defaultzone=cn span=1,1,0,ccs,hdb3 span=2,0,0,esf,b8zs bchan=1-15 dchan=16 bchan=17-31 fxoks=32-55 any clues? p.s. the same setting works fine on HP Proliant server. -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using Modems with Asterisk
Lutgring, Sam wrote: Has anyone had any experience using a modem through the Asterisk system? I have some technical support personnel that need to use a computer modem to connect to a remote system for troubleshooting. Is there a SIP compliant gateway that will support a modem connection at decent speeds (minimum of 28.8) that anyone knows of? If not, has anyone used a Digium FXS card for this? i've done: PRI - Asterisk - SIP to analog adapter - modem or fax and PRI - Asterisk - channel bank - modem or fax they both work fine. -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk crashed
our * crashed twice in a month with segmentation fault
a core dump. here's the stack trace:
#0 0xb7e11965 in mallopt () from /lib/tls/libc.so.6
#1 0xb7e10c43 in malloc () from /lib/tls/libc.so.6
#2 0xb7e17090 in strdup () from /lib/tls/libc.so.6
#3 0x08057ada in ast_verbose (fmt=0x0) at logger.c:879
#4 0xb7b29009 in moh_files_release (chan=0x9455ca0, data=0xb657be48) at
res_musiconhold.c:180
#5 0x080673ae in ast_deactivate_generator (chan=0x9455ca0) at channel.c:1382
#6 0x08068d4e in generator_force (data=0x9455ca0) at channel.c:1405
#7 0x08061c50 in ast_read (chan=0x9455ca0) at channel.c:1857
#8 0x08069293 in ast_generic_bridge (c0=0xb659fcd0, c1=0x9455ca0,
config=0xb6c4feb0, fo=0xb6c4f988, rc=0xb6c4f98c, bridge_end=
{tv_sec = 0, tv_usec = 0}) at channel.c:3260
#9 0x080655fd in ast_channel_bridge (c0=0xb659fcd0, c1=0x9455ca0,
config=0xb6c4feb0, fo=0xb6c4f988, rc=0xb6c4f98c)
at channel.c:3524
#10 0xb78fad29 in ast_bridge_call (chan=0xb659fcd0, peer=0x9455ca0,
config=0xb6c4feb0) at res_features.c:1319
#11 0xb7099301 in dial_exec_full (chan=0xb659fcd0, data=0xb6c4feb0,
peerflags=0xb6c50568) at app_dial.c:1577
#12 0xb7097dc5 in dial_exec (chan=0xb7ed1900, data=0xb7ed1900) at
app_dial.c:1619
#13 0x0808e445 in pbx_extension_helper (c=0xb659fcd0, con=0xb7ed1900, context=0xb659fe20 op05_x, exten=0xb659ff14 00116,
priority=1, label=0x0, callerid=0x0, action=0) at pbx.c:553
#14 0x0808efea in __ast_pbx_run (c=0xb659fcd0) at pbx.c:2227
#15 0x0808fcdf in pbx_thread (data=0xb7ed1900) at pbx.c:2514
#16 0xb7f7cb63 in start_thread () from /lib/tls/libpthread.so.0
#17 0xb7e7718a in clone () from /lib/tls/libc.so.6
another one:
#0 0xb6ff38e2 in decodeMP3 () from /usr/lib/asterisk/modules/format_mp3.so
#1 0xb6ff4be6 in key () from /usr/lib/asterisk/modules/format_mp3.so
#2 0xb6ff4545 in key () from /usr/lib/asterisk/modules/format_mp3.so
#3 0x0806d3a1 in ast_readframe (s=0xb7eb490c) at file.c:570
#4 0xb7b0c134 in moh_files_generator (chan=0xb6b26dc0, data=0xb6b03328, len=0,
samples=160) at res_musiconhold.c:246
#5 0x08068cfe in generator_force (data=0xb6b26dc0) at channel.c:1401
#6 0x08061c50 in ast_read (chan=0xb6b26dc0) at channel.c:1857
#7 0x08069293 in ast_generic_bridge (c0=0xb6b26dc0, c1=0x8699fe8,
config=0xb6677eb0, fo=0xb6677988, rc=0xb667798c, bridge_end=
{tv_sec = 0, tv_usec = 0}) at channel.c:3260
#8 0x080655fd in ast_channel_bridge (c0=0xb6b26dc0, c1=0x8699fe8,
config=0xb6677eb0, fo=0xb6677988, rc=0xb667798c)
at channel.c:3524
#9 0xb78ddd29 in ast_bridge_call (chan=0xb6b26dc0, peer=0x8699fe8,
config=0xb6677eb0) at res_features.c:1319
#10 0xb7033301 in dial_exec_full (chan=0xb6b26dc0, data=0xb6677eb0,
peerflags=0xb6678568) at app_dial.c:1577
#11 0xb7031dc5 in dial_exec (chan=0x48, data=0x48) at app_dial.c:1619
#12 0x0808e445 in pbx_extension_helper (c=0xb6b26dc0, con=0x48, context=0xb6b26f10 op05_x, exten=0xb6b27004 00116, priority=1,
label=0x0, callerid=0x0, action=0) at pbx.c:553
#13 0x0808efea in __ast_pbx_run (c=0xb6b26dc0) at pbx.c:2227
#14 0x0808fcdf in pbx_thread (data=0x48) at pbx.c:2514
#15 0xb7f5fb63 in start_thread () from /lib/tls/libpthread.so.0
#16 0xb7e5a18a in clone () from /lib/tls/libc.so.6
here's the versions of various components:
asterisk: 1.2.7.1, zaptel: 1.2.5, libpri: 1.2.2
any clues would be appreciated?
--
Edwin Lam [EMAIL PROTECTED]
Systems Engineer, Office General, Inc.
Ph: +1 415 439 4988 Fax: +1 415 283 3370
http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20
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Re: [asterisk-users] Caller ID Rewrite
David Bath wrote:
Hi,
Thanks for quick response.
I changed it as you suggested, but it has the same effect:
In the console I get:
--Executing
Set(SIP/604625-b79140a8,CALLERID(number)=44${CALLERID(number)}) in
new stack
It's running the IF code correctly, but in the true it's just not
evaluating the variable...
since the REGEX returns 1 if match. try this instead:
exten=123456,1,Set(CALLERID(number)=
${IF($[REGEX(^0[1-9][0-9]\{9\}$ ${CALLERID(number)}) = 1]?
44${CALLERID(number):1}:${CALLERID(number)})})
Cheers,
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anselm
Martin Hoffmeister
Sent: 01 December 2006 20:26
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Caller ID Rewrite
Am Freitag, den 01.12.2006, 20:03 + schrieb David Bath:
Hi Anselm,
Thanks for the help...
I'm slightly confused as to your response.
Wouldn't that look for a /dialled/ number in the format _0number try
and jump to another extension 0044number with priority 1?
If so, that's not what I'm trying to achieve.
Sorry, my brain is in need for a weekend off work. I obviously
understood your question wrong. My fault.
I've made a bit more progress... and my current diaplan entry looks
like
this:
exten=123456,1,Set(${IF(REGEX(^0[1-9][0-9]\{9\}$
{CALLERID(number)})?CALLERID(number)=44${CALLERID(number):1})})
exten = 123456,2,Goto(sipinternal,101,1)
BUT! There's a very odd problem (and I'm sure it's my fault..) the
second callerid function is not being evaluated...
I _think_ the IF is a string evaluation, so the format should be like
SET MYVARIABLE = [IF condition? value1 : value2]
(see http://www.voip-info.org/wiki/index.php?page=Asterisk+func+if )
Try
exten=123456,1,Set(CALLERID(number)=
${IF(REGEX(^0[1-9][0-9]\{9\}$ ${CALLERID(number)})?
44${CALLERID(number):1}:${CALLERID(number)})})
(two linebreaks to be removed)
HTH,
Anselm
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--
Edwin Lam [EMAIL PROTECTED]
Systems Engineer, Office General, Inc.
Ph: +1 415 439 4988 Fax: +1 415 283 3370
http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20
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Re: [asterisk-users] Question about TFTPD server
Christian wrote: I have installed this package onto my Debian and placed the files i want the Cisco 7960 phone to get from the tftpdboot directory. But it doesn't seem to work. Are ther any special settings I should do to this server? Many thanks for all your help, which tftp server package did you installed? make sure your /tftpboot directory and all the files inside is at least readable by everyone. -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Grandstream GXP2000 -- What's the Catch?
Tom Vile wrote: They brake easy. Speaker phone is not very good. Overall sound not good compared to a Snom, Polycom or Cisco phone. Drop registrations with Asterisk randomly. Power supplies die. Had 4 out of 10 go bad within a year. LCD backlight died on 2 that I deployed. also VLAN doesn't work. (maybe it's fixed in the newest firmware) -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom SP4000 ftp problem
Carla Schroder wrote: On Monday 23 October 2006 17:38, Edwin Lam wrote: Re: [asterisk-users] Polycom SP4000 ftp problem From: Edwin Lam [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Carla Schroder wrote: Sooo...stick with tftp? :) Seriously, that's what it's for. tftp isn't really an FTP server; it uses a different protocol, and uses only a single port (UDP 69). You can't use real FTP servers for this. sure if there's a tftp server that can provide the security and flexibility of ftp server. The difference between tftp and 'real' FTP servers is it does not ask for a login- that's why it's used for diskless clients and PXE net installs. ProFTP (and all other FTP servers) require a login authorization. This is usually invisible to the end-user on public FTP servers, but it's still there. So I'd look for how the phone authorizes itself to the ftp server. the authorization works fine. here's the log from proftpd: 10.1.3.54 UNKNOWN nobody [23/Oct/2006:15:53:48 -0700] USER sp4001 331 - 10.1.3.54 UNKNOWN sp4001 [23/Oct/2006:15:53:48 -0700] PASS (hidden) 230 - 10.1.3.54 UNKNOWN sp4001 [23/Oct/2006:15:53:48 -0700] PWD 257 - 10.1.3.54 UNKNOWN sp4001 [23/Oct/2006:15:53:48 -0700] PASV 227 - 10.1.3.54 UNKNOWN sp4001 [23/Oct/2006:15:53:48 -0700] TYPE I 200 - 10.1.3.54 UNKNOWN sp4001 [23/Oct/2006:15:53:48 -0700] SIZE bootrom.ld 213 - it always stops at the SIZE bootrom.ld mesage. -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom SP4000 ftp problem
Eric ManxPower Wieling wrote: rename bootrom.ld to something else like bootrom.ld-disabled. did that. it hung on sip.ld, rename sip.ld, it hung on phone1.cfg. seems like if the file is bigger than say 1k. it'll hang. -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom SP4000 ftp problem
i recently bought an SP4000 conference phone but having problem provisioning it using ftp, every time it just hangs at Updating initial configuration... screen. when i switch it to tftp, it'll work fine. i though it was bootrom/firmware issue so i upgrade it to bootrom 3.2.2/sip 2.0.1 but it makes no difference. any thoughts? p.s. i'm using debian sarge proftpd 1.2.10 and the setting works fine w/ SP501 with bootrom 3.1.2/sip 1.6.3 -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom SP4000 ftp problem
Carla Schroder wrote: Sooo...stick with tftp? :) Seriously, that's what it's for. tftp isn't really an FTP server; it uses a different protocol, and uses only a single port (UDP 69). You can't use real FTP servers for this. sure if there's a tftp server that can provide the security and flexibility of ftp server. -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] zapata.conf pri
there're several paramters in zapata.conf always puzzled me: pridialplan = prilocaldialplan = internationalprefix = nationalprefix = localprefix = privateprefix = unknownprefix = it only says rarely used for pri in the comments but doesn't explain how to use them. could anyone explain to me how to use these parementers or point me to some documentations on them? thanks. -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] test tone
hi folks. does anybody know what's the phone number for SBC Nothern California's 102-type milliwatt test line? (specifically in 415 area code) -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PRI in Shanghai China
hi folks. does any one have experience setting up E1 PRI in Shanghai, China? it works fine when we use SIP phone to dial out, however when using forward function on the same phone, it seems like it's dialing out but there's actually no respond from the phone company (China Telecom) and eventually the dial command will timed out. here's our PRI portion of zapata.conf: switchtype=euroisdn nsf=none pridialplan=unknown internationalprefix = 00 nationalprefix = 0 localprefix = privateprefix = unknownprefix = signalling=pri_cpe hidecallerid=no usecallingpres=yes callerid=asreceived channel = 1-15 channel = 17-31 any clues? -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Regarding cdr_manager.conf
Victor Alvarez wrote: Hello, My question is.. How does cdr_manager work? Does it suppose to populate cdr-csv/Master.csv? What about the cdr table on the database? What is the event some people talk about? the cdr_manager.conf file control weather the Asterisk manager should include the cdr event. it has nothing to do with Master.csv file. (you can read about Asterisk manager here: http://www.voip-info.org/wiki/view/Asterisk+manager+API ) to enable the cdr engine in general, set enable = yes in cdr.conf and setup at least one of .conf files specific to different databases: Master.csv - cdr_custom.conf Mysql - cdr_mysql.conf odbc - cdr_odbc.conf postgreSQL - cdr_pgsql.conf FreeTDS - cdr_tds.conf -- __ Edwin Lam [EMAIL PROTECTED] __ __ Systems Engineer, Office General, Inc. __ Ph: +1 415 439 4988 Fax: +1 415 283 3370 __ __ http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0xEF11A895 __ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] more voicemail frustrations (was: realtime voicemail)
Vadim Berezniker wrote: That's not a solution, but just a workaround. 1.2.1 has a bug where it always uses an empty context when searching for a mailbox when using realtime config. At around line 546 of apps/app_voicemail.c there is a line that says var = ast_load_realtime(voicemail, mailbox, mailbox, context, retval-context, NULL); Change it to var = ast_load_realtime(voicemail, mailbox, mailbox, context, context, NULL); Then recompile and contexts will work. looks like they've fixed it in 1.2.2. however. i switched over to use realtime SIP. now the voicemail light doesn't work. also has anyone use the MailboxExists() function in dial plan? seems like no matter what i do. it'll just just execute the next proprity. :( -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] linksys SPA-941
does anyone get a hold of the SPA-941 Provisioning Guide? i tried call Sipura's tech support, seems like none of them heard of the term remote provisioning. they kept refering me to their web site which i've check thoroughly, and could not find any documentations on the SPA-941. finally they gave me a phone number to call, which appears to be a fax machine. that's when i gave up on those idiots. -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CDR problem
hi folks. i'm trying to make CDR record the ID of the phone unit that perform a transfer or forward function. but no matter what i do it seems like CDR only record the original caller ID of the call in the src field. for example if someone call in on a zap channel with caller ID '' and destination '', then the user at extension '' transfer the call to ''. the CDR always show '' as the src and '' as the dst. is there any way to change this behavior and have '' as the src? -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dial command in extensions
Kevin Bockman wrote: Patrick wrote: is there anyway to make the dial command return and execute the next line in the dial plan after the channel hangs up? Try with h (for hangup): exten = 1234,1,Dial... exten = 1234,h,... He actually meant the 'h' exten and not priority: exten = h,1,blah yeah. i figured that. but that would execute on everything in the context. Someone else suggested the g option on Dial. that might work better. i'll have to experiment on it. thanks. -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 ___ --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] Dial command in extensions
hi folks. is there anyway to make the dial command return and execute the next line in the dial plan after the channel hangs up? suppose i want to do something like this: exten = 1234,1,dial(SIP/1234) exten = 1234,2,do something but when the dial command hangs up normally, line 2 won't get executed. -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 ___ --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] cisco phones problems
after much struggles. i've found out that if i ping the phone unit from another computer constantly (couple pings every 5-10 sec) the phone will operate fine. once i stopped the pings, the UNREACHABLE message started to pop up and the drop calls problems starts. seems like it's the firmware issue. does anyone uses Cisco SIP 7.3 (or 6.0, i've tried downgraded it at some point) and have similar problems? p.s. another piece of info: the phone units are set to a non default vlan manually since we share the physical lan for both data voice. we recently deployed 10 Cisco 7960G w/ SIP firmware 7.3 on our network and we start having problems of dropping calls (actually the calls wasn't dropped it just the sound was muted for about 5-10 seconds, but most users will think the call dropped and hangup/redial). i've check the console output. there was a lot of messages like the following: Sep 28 15:00:49 NOTICE[8182]: chan_sip.c:6678 handle_response: Peer '3289' is now TOO LAGGED! Sep 28 15:00:59 NOTICE[8182]: chan_sip.c:6672 handle_response: Peer '3289' is now REACHABLE! Sep 28 15:01:08 NOTICE[8182]: chan_sip.c:6678 handle_response: Peer '3201' is now TOO LAGGED! Sep 28 15:01:18 NOTICE[8182]: chan_sip.c:6672 handle_response: Peer '3201' is now REACHABLE! Sep 28 15:04:01 NOTICE[8182]: chan_sip.c:6678 handle_response: Peer '3289' is now TOO LAGGED! Sep 28 15:04:11 NOTICE[8182]: chan_sip.c:6672 handle_response: Peer '3289' is now REACHABLE! Sep 28 15:05:22 NOTICE[8182]: chan_sip.c:8059 sip_poke_noanswer: Peer '3201' is now UNREACHABLE! Sep 28 15:05:32 NOTICE[8182]: chan_sip.c:6672 handle_response: Peer '3201' is now REACHABLE! Sep 28 15:06:23 NOTICE[8182]: chan_sip.c:8059 sip_poke_noanswer: Peer '4881' is now UNREACHABLE! Sep 28 15:06:33 NOTICE[8182]: chan_sip.c:6672 handle_response: Peer '4881' is now REACHABLE! we're running Asterisk 1.0.9 on Debian Sarge w/ custom kernel 2.6.12, strange thing is the system works fine with 2 Cisco phones 8 Grandstream phone before, until i replaced the Grandstreams with Ciscos. the following is typical setting in sip.conf: [1234] context=default type=friend host=dynamic username=1234 secret=test123 mailbox=1234 callerid=John Smith 1234 qualify=yes dtmfmode=rfc2833 any thoughts why this is happening? -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 ___ --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] cisco phones problems
hi folks. we recently deployed 10 Cisco 7960G w/ SIP firmware 7.3 on our network and we start having problems of dropping calls (actually the calls wasn't dropped it just the sound was muted for about 5-10 seconds, but most users will think the call dropped and hangup/redial). i've check the console output. there was a lot of messages like the following: Sep 28 15:00:49 NOTICE[8182]: chan_sip.c:6678 handle_response: Peer '3289' is now TOO LAGGED! Sep 28 15:00:59 NOTICE[8182]: chan_sip.c:6672 handle_response: Peer '3289' is now REACHABLE! Sep 28 15:01:08 NOTICE[8182]: chan_sip.c:6678 handle_response: Peer '3201' is now TOO LAGGED! Sep 28 15:01:18 NOTICE[8182]: chan_sip.c:6672 handle_response: Peer '3201' is now REACHABLE! Sep 28 15:04:01 NOTICE[8182]: chan_sip.c:6678 handle_response: Peer '3289' is now TOO LAGGED! Sep 28 15:04:11 NOTICE[8182]: chan_sip.c:6672 handle_response: Peer '3289' is now REACHABLE! Sep 28 15:05:22 NOTICE[8182]: chan_sip.c:8059 sip_poke_noanswer: Peer '3201' is now UNREACHABLE! Sep 28 15:05:32 NOTICE[8182]: chan_sip.c:6672 handle_response: Peer '3201' is now REACHABLE! Sep 28 15:06:23 NOTICE[8182]: chan_sip.c:8059 sip_poke_noanswer: Peer '4881' is now UNREACHABLE! Sep 28 15:06:33 NOTICE[8182]: chan_sip.c:6672 handle_response: Peer '4881' is now REACHABLE! we're running Asterisk 1.0.9 on Debian Sarge w/ custom kernel 2.6.12, strange thing is the system works fine with 2 Cisco phones 8 Grandstream phone before, until i replaced the Grandstreams with Ciscos. the following is typical setting in sip.conf: [1234] context=default type=friend host=dynamic username=1234 secret=test123 mailbox=1234 callerid=John Smith 1234 qualify=yes dtmfmode=rfc2833 any thoughts why this is happening? -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 ___ --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] cisco phones problems
Greg Oliver wrote: use the qualify= syntax in your sip.conf and make sure it exceeds the latency between the phones and asterisk server in ms. i've set qualify=3000, the unreachable message still popping up occationally. does it seems normal? the server all the Cisco phones are on the same subnet, i can ping them from the operating system with average respond time less than 1 ms. the drop call still happening and i've notice other warning messages on the console: Sep 28 17:08:21 WARNING[8182]: chan_sip.c:695 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Non-critical Request) -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 ___ --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] cisco phones problems
Carlos Alperin wrote: How do you power your cisco phones? Are you using any 35xx XL switch? If that is the case, you need to redo your switch settings enabling QOS. By default CISCO didn't enable it on their switches made for POE the phones. they are all powered with external power supply. we're not using Cisco switches but the server all the phone units are on the same vlan and i've set the QOS priority to highest on our switch for that vlan. -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 ___ --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] Supervised transfer problem with BudgetTone
Nicolas Schmerber wrote: Tried this manipulation a few minutes ago : A calls B , B pushes flash button ( A is waiting with a mp3 played) B calls C pressing Send ; C answers B presses flash button again ; C is so on hold (with a mp3 played) B hangs up But A and C arent in connect ; the phoneof B rings ( to tell someone is in wait : A) So it seems to fail i believe the sequence should be: A call B B pushes flash button (A hears mp3) B calls C pressing send C answers B press transfer (B will hangup, A C is now connected) -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 ___ 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] Asterisk to PSTN
hi folks. i'm planning to connect * to 120 POTS line. i've done some research on FXO cards but unfortunately most manufacturers only make 4 ports/card. the most i've found is 12 ports. so do i have to get 10 of these cards and setup 3 Asterisk servers (assuming each have 4 free PCI slots) link them together with some insane dialplan? or is there an easier way? any suggestions? comments? remarks? parameters? thx. -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 ___ 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] Asterisk to PSTN
Douglas Logan wrote: With 120 Pots lines, why not get 5 T1's, then pick up a couple Digium cards (A Quad T1, and a single T1 card). i'd like to but unfortunately this installation is not in the US and we have to keep the 120 phone numbers (which are not sequential) porting over those numbers to E1 lines w/ DID seems out of the question. i guess the way to go is using channel banks to convert those to E1 then connect Asterisk that way. further research, how about using these: http://www.welltech.com.tw/product_e_03.htm will that work? -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 ___ 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] Sipura SPA-841
does anybody has experienece with Sipura SPA-841 phone unit? how's its sound quality especially speaker phone? i have several Grandstream phones and was getting fustrated about the quality and bugs of their firmware. -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 ___ 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] connecting Asterisk to NEC NEAX system
hi. i connected Asterisk to an NEC NEAX system with a crossover T1 cable and the Digium TE405P using EM wink signaling. the connection's ok. however when dialing from the NEC to the Asterisk. most of the time the Asterisk only sees the first digit of the dialed number(which is 4 digits). some time if i dialed the 4 digits very fast it might get through. seems like there's a timming issue of the DTMF. what can i do to solve this? i looked through the docs for zaptel.conf zapata.conf and doesn't seems there's any parameters to control the DTMF timing. -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 ___ 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] connecting Asterisk to NEC NEAX system
Dennis Walker wrote: in zapata.conf use emdigitwait=### number of milliseconds to wait for digits to be output I had similar problems with MITEL system and had emdigitwait=500 aha.. thanks. incidentally i'm using the stable 1.0.7 version. so i have to manually patch chan_zap.c. also i have to set emdigitwait=1000 to get it to work reliably. -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 ___ 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
