[Asterisk-Users] Two devices behind nat
Hello Everyone, I have one machine (asterisk server) that is DMZ behind my nat firewall on my client end (at home) i have a linksys wrt54g with 16384-32766 forwarded to my cisco 7960 (which works fine) and 16000 - 16383 forwarded to my sipura 2100 (which is set to these ports) For some reason, the sipura only works some of the time. Both devices are set to register, is there anything else I need to do to get them to work behind nat? Thanks! Chris ___ --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] error messages
Hey, does anyone know why i'd be receiving: Aug 28 19:40:04 DEBUG[1875]: # Testing 66.27.233.241 with 10.0.10.0 Aug 28 19:40:04 DEBUG[1875]: Target address 66.27.233.241 is not local, substituting externip I get tons of them, usually when the phone is registering/calling/receiving calls. Thanks! Chris ___ --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] Flash over SIP Trunk
Hello, I have a Incoming/Outgoing SIP Trunk setup to Broadvoice, is there a way to send a "Flash" over the trunk, for example, to do flash transfers and call-waiting? I tried to use Flash() but it seems to not work on the sip trunk, only my zap trunks. Please let me know, thanks! :) Chris ___ 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 Flash Transfer (callthrough)
Hello Everyone:)!,
I have a Incoming/Outgoing SIP Trunk setup to Broadvoice, is there a way to
send a Flash over the trunk, for example, to do flash transfers and
call-waiting?
I tried to use Flash() but it seems to not work on the sip trunk, my
configuration is as follows:
exten = 500,1,Goto(callthrough,s,1)
[callthrough]
exten = s,1,SetVar(NR=)
exten = s,2,Background(privacy-prompt)
exten = s,3,ResponseTimeout(10)
exten = s,4,WaitExten(20)
exten = _X,1,SetVar(NR=${NR})
exten = _X,2,Goto(s,3)
exten = *,1,Goto(s,1)
exten = #,1,Playback(transfer)
exten = #,2,Flash()
exten = #,3,SendDTMF(${NR})
exten = #,4,Flash()
exten = #,5,Hangup()
Debug output is as follows:
Please let me know, thanks! :)
Chris
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[Asterisk-Users] Routing by Called interface
Hey everyone, I want to run different lines directly to different extensions on two FXO analog interfaces. ie; Zap/1 goes to Ext. 101, Zap/2 goes to extensions 102 Does anyone know of a way to do this? Thanks! Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Routing by called interface
Hey everyone, I want to run different lines directly to different extensions on two FXO analog interfaces. ie; Zap/1 goes to Ext. 101, Zap/2 goes to extensions 102 Does anyone know of a way to do this? Thanks! Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DNIS on X100P
Hello, I have 6 analog lines that ring downcoming into my office that support DNIS, my current phone system (SRX)displays the "called number" on the screen of the operator phone, IE; xxx-7873 = Netxn, xxx-7874 = Dolphinsafe, etc. Does asterisk support any type of features to distinguish between the numbers dialed? Thanks! Chris Wilson
[Asterisk-Users] Cisco 7960 Problems
Has anyone ever seen these errors generated by a cisco 7960? none of our other brand phones seem to generate these erros: Jan 27 21:54:07 WARNING[-1147556944]: chan_sip.c:486 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 101 (Response)Jan 27 21:54:08 WARNING[-1147556944]: chan_sip.c:486 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 101 (Response)Jan 27 21:54:12 WARNING[-1147556944]: chan_sip.c:486 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 101 (Response)Jan 27 21:54:14 WARNING[-1147556944]: chan_sip.c:2485 __transmit_response: Unable to determine sequence number from ''Jan 27 21:54:18 WARNING[-1147556944]: chan_sip.c:2485 __transmit_response: Unable to determine sequence number from ''Jan 27 21:54:22 WARNING[-1147556944]: chan_sip.c:2485 __transmit_response: Unable to determine sequence number from '' Thanks! Any feedback would be appreciated :) Chris
Re: [Asterisk-Users] Re: retrans_pkt: Maximum retries exceeded on call
Hmm, The host seems to be good, I have no firewall rules in place at the moment for the local network, and everything is consistantly reachable. it seems to only happen when a call is hung up/initiated, and when the program is first started...if that might provide any insight. Thanks!:) Chris - Original Message - From: Doug Meredith [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, January 24, 2004 4:02 PM Subject: [Asterisk-Users] Re: retrans_pkt: Maximum retries exceeded on call Chris Wilson [EMAIL PROTECTED] wrote: Hey, I'm getting an odd message in my logs, and have'nt been able to find much information on it: Jan 24 00:22:39 WARNING[-1137431632]: chan_sip.c:486 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Request) Just guessing here, but it sounds like Asterisk sent a request, didn't get a reply, sent again, didn't get a reply, and so on until it hit an internal limit. If my guess is correct, I suppose there could be many causes, including: * Target host down * No path to the target * Firewall blocking traffic * Target host not running SIP, at least on the targeted port. Doug -- Doug Meredith ([EMAIL PROTECTED]) SystemGuard - Oracle remote support 877-974-8273 (87-SYSGUARD) 506-854-7997 www.systemguard.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] SayDigits
Awesome, that worked! Thanks :)
Chris
- Original Message -
From: Grzegorz Nosek [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, January 25, 2004 1:31 AM
Subject: Re: [Asterisk-Users] SayDigits
On Sat, 24 Jan 2004 10:56:59 -0800, Chris Wilson wrote
Has anyone had this problem:
(When calling to ext. 1010)
Jan 24 10:50:27 WARNING[-1252262992]: file.c:446
ast_openstream: File digits/ does not exist in any format
Jan 24 10:50:27 WARNING[-1252262992]: file.c:734
ast_streamfile: Unable to open digits/ (format ULAW): No
such file or directory
in Extensions.conf
exten = 1010,1,SayDigits(${CALLERID})
/var/lib/asterisk/sounds/digits exists, and there are many
files in there. Any idea's?
Thanks! :)
Chris
have you tried:
exten = 1010,1,SayDigits(${CALLERIDNUM})
?
hth,
greg
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[Asterisk-Users] retrans_pkt: Maximum retries exceeded on call
Hey, I'm getting an odd message in my logs, and have'nt been able to find much information on it: Jan 24 00:22:39 WARNING[-1137431632]: chan_sip.c:486 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Request) I'm running asterisk with a Cisco 7960G If anyone know's why i'd get this.Any help would be appreciated! =] Thanks! Chris
Re: [Asterisk-Users] retrans_pkt: Maximum retries exceeded on call
Hm, had that enabled since i set everything up. I tried with nat=no as well, same problem. Welp, I guess if anyone figgers it out i'd appreciate any help that comes my way :). Thanks! Chris - Original Message - From: Kannaiyan Natesan To: [EMAIL PROTECTED] Sent: Saturday, January 24, 2004 12:53 AM Subject: Re: [Asterisk-Users] retrans_pkt: Maximum retries exceeded on call You are having Cisco 7960G behind NAT. Try with nat=yes I'm not sure any other settings will solve that in asterisk. I have tried but no luck. Kannaiyan - Original Message - From: Chris Wilson To: [EMAIL PROTECTED] Sent: Saturday, January 24, 2004 8:26 AM Subject: [Asterisk-Users] retrans_pkt: Maximum retries exceeded on call Hey, I'm getting an odd message in my logs, and have'nt been able to find much information on it: Jan 24 00:22:39 WARNING[-1137431632]: chan_sip.c:486 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Request) I'm running asterisk with a Cisco 7960G If anyone know's why i'd get this.Any help would be appreciated! =] Thanks! Chris
[Asterisk-Users] SayDigits
Has anyone had this problem:
(Whencalling to ext. 1010)
Jan 24 10:50:27 WARNING[-1252262992]: file.c:446
ast_openstream: File digits/" does not exist in any formatJan 24 10:50:27
WARNING[-1252262992]: file.c:734 ast_streamfile: Unable to open digits/" (format
ULAW): No such file or directory
in Extensions.conf
exten =
1010,1,SayDigits(${CALLERID})
/var/lib/asterisk/sounds/digits exists, and
there are many files in there. Any idea's?
Thanks! :)
Chris
Re: [Asterisk-Users] Mute button in Grandstream?
Hi John, On Fri, 28 Nov 2003, Anton Yurchenko wrote: Hello, Has anybody been able to get the Mute button work on grandstream? it simply does nothing. Only Hold is avalable, which is not that good. Does the GS even HAVE a mute button? The 101's appear not to. My 100 has MUTE/DEL in the bottom right hand corner. And no, it doesn't work for me either :-) Cheers, Chris. -- _ __ __ _ / __/ / ,__(_)_ | Chris Wilson -- UNIX Firewall Lead Developer | / (_ ,\/ _/ /_ \ | NetServers.co.uk http://www.netservers.co.uk | \__/_/_/_//_/___/ | 21 Signet Court, Cambridge, UK. 01223 576516 | ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 3x AVM Fritz!Card PCI for a EuroISDN PBX.
Hi David, I'm currently considering various PBX solutions for our office telephone network, and would very much like to use Asterisk. Currently, my research is incomplete. I have been recommended to use the above cards, but it is unclear from my Googling whether my configuration will work: - 3x Fritz!Card PCI's in one host. - 3x 6 b-channels. - ~20 Budgetone (and some others) handsets. Can anyone answer these questions: - Will the 3 ISDN cards function correctly in one host? We have done this and it works, using a variant of the hack posted on the website. Points to watch out for: - It's not reliable. We've had Asterisk spontaneously refuse to dial out or accept connections on CAPI until the cards are reset. We don't recommend doing this in production. - You need to be careful with the patch because there are two types of cards, and the patch isn't clever about how it detects them, so either make sure that all your cards are absolutely identical in /proc/pci, or fix the patch. - Will running all 3 cards flat out require particularly beefy hardware? Doesn't seem to. - Will the Grandstream phones provide a good equivilant to professional dedicated PBX phones? (assuming a good network) I have read lots about echo problems and so on, is this an issue? They are cheap and nasty feeling, and not particularly reliable, so I would say no. Cisco 7960 is much better, although more of a pain to get working out of the box, since you need DHCP, TFTP and configuration tools. Cheers, Chris. -- _ __ __ _ / __/ / ,__(_)_ | Chris Wilson -- UNIX Firewall Lead Developer | / (_ ,\/ _/ /_ \ | NetServers.co.uk http://www.netservers.co.uk | \__/_/_/_//_/___/ | 21 Signet Court, Cambridge, UK. 01223 576516 | ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Larger SIP packets
Hi all, We would like to increase the size (sample length) of RTP packets sent by Asterisk to SIP phones. I gather that Asterisk currently always uses 20ms packets for RTP, although I can't find in the source where that's defined, unless it's in chan_zap.c. I'm guessing from [http://lists.digium.com/pipermail/asterisk-users/2003-September/019490.html] that it's not possible right now (or at least wasn't in September). Does anyone know if and when it's coming as an option, or if we can modify the source to always use, say, 40ms packets instead of 20? Thanks in advance for any advice. Please CC me since I'm not subscribed to the list anymore (too much traffic). Cheers, Chris. -- _ __ __ _ / __/ / ,__(_)_ | Chris Wilson -- UNIX Firewall Lead Developer | / (_ ,\/ _/ /_ \ | NetServers.co.uk http://www.netservers.co.uk | \__/_/_/_//_/___/ | 21 Signet Court, Cambridge, UK. 01223 576516 | ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ISDN PBX + IVR + Voicemail Configuration - Sanity Check ...
Hi Hans, b. Hooked up to ISDN PBX using an ISDN4Linux (chan_modem ?) or CAPI4Linux (chan_capi ?) compatible ISDN controller. We had real problems with chan_modem, where Asterisk would deadlock after the first call was hung up. We saw this with both 0.5.0 and CVS from late October. Other people have reported success with chan_modem, so I don't know if it's a site-specific issue. We are using CAPI now which seems to work reasonably well, except that we had an unexplained total ISDN failure a few days ago, which was fixed by restarting Asterisk. In case anyone is interested in the deadlock, which also affects analog modems for us, here is the backtrace: #0 0xe002 in ?? () #1 0x08052355 in ast_log (level=0, file=0x80b42a8 asterisk.c, line=337, function=0x80b4304 urg_handler, fmt=0x80b42f4 Urgent handler\n) at logger.c:266 #2 0x080832a2 in urg_handler (num=-4) at asterisk.c:337 #3 signal handler called #4 0xe000 in ?? () #5 0x08093ef6 in el_getc (el=0x80ca970, cp=0xbfffea4b @øê\f\b\214ê^?¿\002) at read.c:347 #6 0x08093d65 in read_getcmd (el=0x80ca970, cmdnum=0xfffc Address 0xfffc out of bounds, ch=0xbfffea4b @øê\f\b\214ê^?¿\002) at read.c:243 #7 0x08094025 in el_gets (el=0x80ca970, nread=0xbfffea8c) at read.c:443 #8 0x08082761 in main (argc=135064312, argv=0xbfffea8c) at asterisk.c:1459 #9 0x420156a4 in __libc_start_main () from /lib/tls/libc.so.6 (gdb) (gdb) up #1 0x08052355 in ast_log (level=0, file=0x80b42a8 asterisk.c, line=337, function=0x80b4304 urg_handler, fmt=0x80b42f4 Urgent handler\n) at logger.c:266 266 ast_mutex_lock(loglock); P.S. He who comes up with clean internal ISDN bus (point to multi-point) support for Asterisk, based on CologneChip based equipment receives an 18 large Dutch cheese in the mail, right after I've wiped away my tears of happiness ! We are really looking forward to trying this too. Klaus-Peter is the man =) (sorry dan, can't use your softphone on *nix). Cheers, Chris. -- ___ __ _ / __// / ,__(_)_ | Chris Wilson -- UNIX Firewall Lead Developer | / (_ / ,\/ _/ /_ \ | NetServers.co.uk http://www.netservers.co.uk | \ _//_/_/_//_/___/ | 21 Signet Court, Cambridge, UK. 01223 576516 | ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: *, Fritz!PCI and strange behavior
Hi Patrick and others, On Tue, 4 Nov 2003, Patrick Lidstone (Personal E-mail) wrote: - Very often, after * runs for a while, it stops recognizing incoming ISDN calls and refuses to send out ISDN calls. I have this. If I try to dial out, I get an all channels are busy at this time error, when they are not. How did you get this error displayed? All I get with CAPI DEBUG is a reason code. The funny thing is, restarting * or CAPI doesn't work - I have to shutdown both, unplug and replug the ISDN cable, and then after startup everything works again. At first, I thought that it might be a bad cable, so I taped down everything in order to prevent it from moving. This didn't help. I really do not understand why the thing with the cable is necessary. Straight asterisk restart always clears this condition for me. We just experienced a problem which looked something like this. Nobody could call in or out through the Fritz cards until we restarted Asterisk. The CAPI debug output looked like this: == CONNECT_IND (PLCI=0x101,DID=510,CID=221,CIP=0x1,CONTROLLER=0x1) == DISCONNECT_IND PLCI=0x101 REASON=0 == CONNECT_IND (PLCI=0x101,DID=510,CID=(null),CIP=0x4,CONTROLLER=0x1) == DISCONNECT_IND PLCI=0x101 REASON=0 where CONNECT_IND was immediately followed by DISCONNECT_IND. There was none of the normal output: -- data = 514:901482320681 -- capi request omsn = 514 == found capi with omsn = 514 or CONNECT_CONF/CONNECT_B3_REQ/DISCONNECT_B3_IND. Reason code 0 looks very suspicious. /var/log/messages showed: Nov 6 13:11:42 voip kernel: kcapi: appl 1 ncci 0x10102 up Nov 6 13:12:19 voip kernel: kcapi: appl 1 ncci 0x10102 down Nov 6 13:13:35 voip kernel: kcapi: appl 1 ncci 0x10101 up Nov 6 13:13:41 voip kernel: kcapi: appl 1 ncci 0x10101 down Nov 6 13:13:49 voip kernel: kcapi: appl 1 ncci 0x10101 up Nov 6 13:13:58 voip kernel: kcapi: appl 1 ncci 0x10101 down Nov 6 13:15:26 voip kernel: kcapi: appl 1 ncci 0x10101 up Nov 6 13:15:56 voip kernel: kcapi: appl 1 ncci 0x10101 down Does anybody have any ideas? Any light that you can shine on this would be most helpful. OBTW: the answer don't use a Fritz is not applicable here - I'm trying to assess the feasibility of making a 300$ ISDN SoHo PBX... I think the problem may be related to call progress indication from the ISDN line. I have UK ISDN2e (packaged as Business Highway - which includes what is effectively a telco-owned TA with two analogue ports). I have noticed that outgoing channels getting tied up corresponds to placing a call which is terminated prematurely (e.g. hangup before completing dialing) or dialing a call which can't be completed because the dialed PSTN subscriber number is invalid. I've learned to live with it - but it would be great to get to the bottom of it. Sorry, forgot to run show channels and capi info. Will try to remember next time. Cheers, Chris. -- ___ __ _ / __// / ,__(_)_ | Chris Wilson -- UNIX Firewall Lead Developer | / (_ / ,\/ _/ /_ \ | NetServers.co.uk http://www.netservers.co.uk | \ _//_/_/_//_/___/ | 21 Signet Court, Cambridge, UK. 01223 576516 | ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] NetJet Cards
Hi Matthew,
On 2 Nov 2003, Matthew Enger wrote:
exten = _004,1,Dial(modem/g1/V${EXTEN:1})
Try this Dial command:
Dial(Modem/ttyI0:${EXTEN:1})
msn=0397468733L*
Try removing L* from the MSN, it looks wrong to me.
You might find that ttyI0 and ttyI1 are both channels of the first card,
and that you will need ttyI2 and ttyI3 for the second card. But I haven't
tested isdn4linux with more than one card.
Please let me know if you get it working, I had major problems with
Asterisk hanging using the isdn4linux and other modem drivers. Eventually
I had to switch to CAPI.
Cheers, Chris.
--
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