Re: [asterisk-users] call files
I found the decision in using
Channel: Local/[EMAIL PROTECTED]/n
Denis V. Gudtsov пишет:
Hello, All!
How to specify the context in call file section Channel? Is it possible?
I want to dial external number (12345) and connect it to context
notify, which consist of playback() command:
Channel: SIP/12345
Callerid: auto 12345
MaxRetries: 3
RetryTime: 40
WaitTime: 50
Context: notify
Extension: 1
Priority: 1
extensions.ael follows:
context notify {
1 = {
start:
Answer();
Wait(1);
Playback(ulii_01);
HangUp();
};
I want to dial number 12345 with taking into account the dial plan,
written in context.
when i'm trying to set:
Channel: SIP/[EMAIL PROTECTED]
asterisk say's:
chan_sip.c:2737 create_addr: No such host: context
attempt to set:
Channel: SIP/context/12345
has the same result
asterisk version is 1.4.2
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[asterisk-users] How to return dialstatus of second (sub) call
Hello all
I have this problem, i need a way to balance my trunks which are SIP
peers, when a SIP peer is busy then send the call for another peer and
so until i can send away the call, i think i can do it with queues.
Ok this is the scenario:
In extensions.conf
[balance]
exten = _,1,NoOp(Call to: ${EXTEN})
exten = _,2,Answer()
exten = _,3,SetVar(_ORGEXTEN=${EXTEN})
exten = _,4,SetVar(_ORGUNIQUEID=${UNIQUEID})
exten = _,5,Set(CDR(userfield)=${ORGUNIQUEID})
exten = _,6,Queue(qtest,r)
exten = _,7,Hangup()
I have a queue with 100 members which are local channels
In queues.conf
[qtest]
strategy=random
member=Local/[EMAIL PROTECTED]
member=Local/[EMAIL PROTECTED]
member=Local/[EMAIL PROTECTED]
...
...
member=Local/[EMAIL PROTECTED]
Each member is an extension.
In extensions.conf
[salidas]
exten = 1,1,Dial(SIP/TRUNK1/${ORGEXTEN},60,r)
exten = 2,1,Dial(SIP/TRUNK2/${ORGEXTEN},60,r)
exten = 3,1,Dial(SIP/TRUNK3/${ORGEXTEN},60,r)
...
...
exten = 100,1,Dial(SIP/TRUNK100/${ORGEXTEN},60,r)
exten = h,1,Set(CDR(userfield)=${ORGUNIQUEID})
Each TRUNK is a SIP peer. And i have anothers users which have the
context balance.
The question now is that i need return on the SIP message the
dialstatus of the Dial on salidas context to the peer which is doing
the call, now i always return ANSWARED because the Queue always
answare, i need some way to tell the queue not return the dialstatus
until the second call finish and i know the dialstatus and then return
it dialstatus to the peer which is doing the call in the SIP message.
Something like this
User --- doing the call Queue
|
-- Trying
|
-- Dial on salidas
|
-- Finish the call
|
Return the dial status--
Someone have an idea?
Regards.
--
Jonathan Alberto Rivera Gomez
http://linuxuanl.org
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Re: [asterisk-users] Red alarms
On 2/8/07, Wayne Jensen [EMAIL PROTECTED] wrote: On 2/8/07, Don Pobanz [EMAIL PROTECTED] wrote: Asterisk is getting red alarms on my T1, sometimes once or twice a day, but today it happened 5 times. Even once is too many. Every call in progress is dropped. Red alarm means that the hardware is not seeing the T1 signal coming in. This most likely is a cable or wiring or perhaps a hardware problem. I know that it's not wiring because it's never happened before in the year and a half that we've been using it, and if I disconnect the asterisk box and use it the same way we've been using it, the problem goes away. same cabling, just unplugging the cable from the channel bank and plugging it into the Digium card. does this mean that the Digium card is bad? or is there something in the configuration that could make it not see the T1 signal coming in? I don't see how it would be, but is it possible that a bad cable between the Digium card and the channel bank (channels 49-72 on the card) could cause a red alarm on channels 1-24? I've used the same Digium card in two different boxes and got the same red alarms, so I doubt it's a problem with the computer it's in. The red alarms only happen when there's a high volume of calls going through, but it doesn't *always* happen when there's a high volume. Plugging another T1 into the card seems to have cured it... strange. When I was getting all the red alarms, I had one cable coming into the card (channels 1-24) and two cables going out (channels 49-96), leaving one slot (channels 25-48) empty. Once I plugged another T1 into that slot, the red alarms pretty much went away. Why would that happen? ___ --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] SDP bug
3 apr 2007 kl. 09.07 skrev Raj Jain: Olle, It depends on how strictly the UA adheres to the offer/answer model. The issue would be that a RE-INVITE from Asterisk will have the version number incremented by more than one, which will break the following rule. Quoting from RFC 3264 Section 8: When issuing an offer that modifies the session, the o= line of the new SDP MUST be identical to that in the previous SDP, except that the version in the origin field MUST increment by one from the previous SDP. That said, I agree that most UAs do not check this. What's a bit more alarming fundamentally is that Asterisk is creating a new answer SDP to respond to an INVITE retransmission. An RFC 3261 compliant implementation MUST send an exact copy of the previous SIP response. Anyway, I realize that Asterisk is not inherently RFC 3261 compliant. Well, the whole retransmit engine is flawed in Asterisk, something I will try to fix in pineapple, the project I'm trying to start as a major rework of the SIp channel. See http://www.codename-pineapple.com. The project is stalled due to lack of funding. I have a few sponsors - thank you! - but not enough to dedicate my time for it. However, this thing about the SDP seems like something that doesn't really disturb communication today, even though I admit we're doing it wrong. At this point, we need to focus on fixing bugs that make communication and interoperability impossible, after that we can fix issues like this that are wrong, but isn't proved to have an affect on interoperability or communication. Gotta focus :-) /O ___ --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] SDP bug
3 apr 2007 kl. 10.04 skrev kjcsb: The call that gets dropped had a retransmission of INVITE from UAC to UAS (and therefore retransmission of 200 OK from UAS to UAC). There is nothing wrong with the re-transmission as such, but I noticed a potential bug in Asterisk in the way it responds to an INVITE retransmission. Asterisk is bumping up the session version number in the retransmitted 200 OK's SDP. This is as if Asterisk is treating the INVITE retransmission as a RE-INVITE. Asterisk sends 200 OK: o=root 16300 16300 IN IP4 203.89.nnn.nnn Asterisk sends 200 OK (retransmission): o=root 16300 16301 IN IP4 203.89.nnn.nnn Ideally, this bug should have nothing to do with why Asterisk is ignoring the ACK (which is why it keeps reatrasmitting the 200 OK and eventually drops the call). However, if you can confirm that all dropped calls have INVITE retransmission then that might give us a clue? Raj, That's an interesting observation. Do you think this will cause any issues? Even though it's not beautiful, I fail to see why a UA would check that. I have run a number of tests and in all cases the calls that fail have a retransmitted INVITE whereas the successfull calls have only one INVITE. I need to see a full SIP debug to check what's going on. /O ___ --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] Re: asterisk-users Digest, Vol 33, Issue 20
Je suis absent du 2/04/2007 au 11/04/2007. Je répondrai à votre message dès mon retour. Pour toute urgence, contacter Emmanuelle Parache Moga ou Cédric Buzay. ___ --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] Asterisk server hangs on after only few hours again.
johnny_xing wrote: hi, everyone, i have been sufferred for the asterisk hang on problem for so long and i just reinstalled the whole thing yesterday, but again this morning the server hangs on again, you could not call in through PSTN line and the ppl also could not call out throught the server, there is simply engaged dial tone when you try to do so. and the only thing i can do is to restart asterisk server after some hours or one day. i am using asterisk 1.2.17 + zaptel 1.2.16 + freepbx 2.2.1. Any one please give me some advice on this? thanks so much really, or how I can monitor and debug the problems when I happened again next time. My guess would be that you have a hardware issue. Either a bad piece of hardware or a hardware compatibility issue. Check the output of cat /proc/interrupts to make sure you don't have any IRQ sharing. I have personally not had good luck with Digium analog cards, but most people seem to use then without issue. ___ --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] SNOM and Got SUBSCRIBE for extensions without hint. Please add hint to *8 in context inbound_sip
Hi, I have an Asterisk 1.2.9.1 box and a bunch of snom phones. I sometimes get this error: *ERROR[31201] chan_sip.c: Got SUBSCRIBE for extensions without hint. Please add hint to *8 in context inbound_sip* It seems that my SIP phone is sending subscribe command for numbers not inserted inside the snom function keys (and *8 surely is not). I searched on internet and Snom documentation but found nothing to avoid the Snom to send this commands. Is there anybody who knows how to? TIA Giorgio ___ --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] Re: Correct latency values in sip show peers
Eric ManxPower Wieling wrote: The times shown are the time to get a response to a SIP OPTIONS packet sent to the phone, not the time to get a response from an ICMP ECHO (ping) packet. What's the difference between yours and mine mail? -- Tomislav Parcina [EMAIL PROTECTED] ___ --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 1.2.17 and BRIstuff - SOLVED
Hi All, I'm using Asterisk 1.2.17 and I would like to install BRIstuff. Problem is that current version of BRIstuff is for Asterisk 1.2.14. BRIstuff 0.3.0-PRE-1y (* 1.2.14) If I'm misunderstanding how to apply patches for 1.2.17? Thank you in advance, Dominik I found it:) http://www.junghanns.net/downloads/bristuff-0.3.0-PRE-1y-e.tar.gz Regards, Dominik ___ --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 1.2.17 and BRIstuff
Hi All, I'm using Asterisk 1.2.17 and I would like to install BRIstuff. Problem is that current version of BRIstuff is for Asterisk 1.2.14. BRIstuff 0.3.0-PRE-1y (* 1.2.14) If I'm misunderstanding how to apply patches for 1.2.17? Thank you in advance, Dominik ___ --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] Asterisk 1.2.17 and BRIstuff
Dominik Zalewski wrote: I'm using Asterisk 1.2.17 and I would like to install BRIstuff. Problem is that current version of BRIstuff is for Asterisk 1.2.14. BRIstuff 0.3.0-PRE-1y (* 1.2.14) If I'm misunderstanding how to apply patches for 1.2.17? Download Bristuff 0.3.0-PRE-1y-e: http://www.junghanns.net/downloads/bristuff-0.3.0-PRE-1y-e.tar.gz and ./install.sh It will download Asterisk 1.2.17. No need to manually apply any patches. Regards, Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --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] SDP bug
Olle, Regarding project Pineapple, I'm curious why rewrite (or refactor) the SIP stack instead of using an open-source one. Did your research show that there is nothing viable out there that'll fit well w/in Asterisk? OpenPBX community is talking about using Sofia-SIP stack, for instance. Raj Well, the whole retransmit engine is flawed in Asterisk, something I will try to fix in pineapple, the project I'm trying to start as a major rework of the SIp channel. See http://www.codename-pineapple.com. The project is stalled due to lack of funding. I have a few sponsors - thank you! - but not enough to dedicate my time for it. ___ --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] remote SIP, no audio, or one way audio.
Joe Acquisto wrote: Thanks. And this might go where, in rc.d/rc.firewall.local ? But I don't get it. Isn't this redundant? Since I have port forwarding already. . .? joe a. ___ --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 What this is doing is allowing unfettered access between your PBX and phones. Too many people forget that a VoIP transaction consists of more than just opening up ports 5060 and 5061. This are used for registration/administration, etc., in the case of one way audio, or audio for any matter, this is carried out by RTP on separate ports which will never be the same port unless you have it specified. Summarized: NAT + VoIP = nightmare If at all doable, segment your phones out to a DMZ with VLANs, constructive routing, and ACL's to avoid leveraged security incidents via those phones being opened. http://www.voip-info.org/wiki/index.php?page=RTP+Symmetric http://www.voip-info.org/wiki/view/NAT+and+VOIP -- J. Oquendo http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x1383A743 sil . infiltrated @ net http://www.infiltrated.net The happiness of society is the end of government. John Adams smime.p7s Description: S/MIME Cryptographic Signature ___ --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] remote SIP, no audio, or one way audio.
J. Oquendo [EMAIL PROTECTED] Wrote: 4/5/2007 6:47 AM: Joe Acquisto wrote: Thanks. And this might go where, in rc.d/rc.firewall.local ? But I don't get it. Isn't this redundant? Since I have port forwarding already. . .? joe a. ___ --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 What this is doing is allowing unfettered access between your PBX and phones. Too many people forget that a VoIP transaction consists of more than just opening up ports 5060 and 5061. This are used for registration/administration, etc., in the case of one way audio, or audio for any matter, this is carried out by RTP on separate ports which will never be the same port unless you have it specified. Summarized: NAT + VoIP = nightmare If at all doable, segment your phones out to a DMZ with VLANs, constructive routing, and ACL's to avoid leveraged security incidents via those phones being opened. Thanks. Do you have recommended switches, capable of supporting VLAN's in an appropriate manner? The cheaper the better, at this point. I have attempted VLAN's several times, for this purpose specifically, using Nortel Baystack 450-24's. Not working as one would expect. Some say these simply do not do VLAN's properly This can go off list, if it is OT. joe a. ___ --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 repair
James Andrewartha, We send these into Polycom for repair on occasion. It's will cost roughly $200~$300 direct with Polycom and will take 8~10 weeks for repair. Obviously, most people opt not to do a repair and just buy a new phone instead. The only time it may make sense to repair an out of warranty phone would be if you have some sentimental attachment to your current phone or more realistically, you're not certain the newer phones will work with your PBX or SP. There may be other reasons, but I can't think of those right now. Keep in mind, IP 600's aren't available any longer from Polycom. The new replacement is the IP 601 phone or the IP 650. Jessee Holmes Atacomm / Ataractic Corporation www.atacomm.com V: 1-877-700-VOIP [EMAIL PROTECTED] Looking for voice over IP products? Visit our VoIP store at http:// voipstore.atacomm.com/ On Apr 4, 2007, at 11:36 PM, James Andrewartha wrote: Hi all, Has anyone had any experience getting Polycom phones repaired? The screen on one of our IP600s got smashed, and I'm wondering if it's worth the effort to get it repaired, or if it'd just be cheaper to buy a new phone. Thanks, -- James Andrewartha Systems Administrator Data Analysis Australia Pty Ltd ___ --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 ___ --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 DCHAN Errors
Hey all, I had a user complaining of calls which were dropping mid-conversation. I looked into the time of one of the calls, and saw the following: Apr 4 12:13:03 WARNING[6670] chan_zap.c: No D-channels available! Using Primary channel 28 as D-channel anyway! Apr 4 12:13:05 WARNING[6660] channel.c: Avoided initial deadlock for '0x82b8430', 10 retries! Apr 4 12:13:05 WARNING[6660] channel.c: Avoided initial deadlock for '0x82d9920', 10 retries! Apr 4 12:13:05 WARNING[6660] channel.c: Avoided initial deadlock for '0x82369f0', 10 retries! Apr 4 12:13:05 WARNING[6660] channel.c: Avoided initial deadlock for '0x8242968', 10 retries! Apr 4 12:13:06 WARNING[6670] chan_zap.c: No D-channels available! Using Primary channel 28 as D-channel anyway! Apr 4 12:13:09 NOTICE[15466] app_dial.c: Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion) Apr 4 12:13:11 NOTICE[6670] chan_zap.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 2 I have read that some people believe this is a driver problem, which others believe something could be wrong with the PRI itself. I did a zttool and there are and haven't been any red alarms. I also read some people believe this issue can be caused by another card in the box taking too long to respond, such as an IDE card. We have 2 sangoma PRI cards and 1 sangoma FXO/FXS card. Any thoughts? ___ --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] SDP bug
5 apr 2007 kl. 13.04 skrev Raj Jain: Regarding project Pineapple, I'm curious why rewrite (or refactor) the SIP stack instead of using an open-source one. Did your research show that there is nothing viable out there that'll fit well w/in Asterisk? OpenPBX community is talking about using Sofia- SIP stack, for instance. Well, tests show that ours isn't that bad from an interoperability standpoint. Two years ago, I was convinced that it was the only way to go. I still haven't ruled it out though. I admit that there are a lot of picky stuff we don't support, but so far we interoperate with most products out there successfully. It's way better now than when I started working with chan_sip. I have been in contact with many SIP stack developers, and there are things we need to do that is hard to do without having to change their stack. And also, many stacks have incompatible licenses, which rules them out. Regardless, pineapple is much more than the lower layer stack. Check http://www.codename-pineapple.org and you will see that it's a lot about changing the concepts and how we interface SIP to the core PBX too. Due to lack of enough sponsors in the community, I have to postpone the project. Hopefully I can get it going later this year, but right now I have to focus on other projects. If there are interested sponsors out there, please send me an e-mail. /O ___ --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] Call dies when I press *
Is it related to Dial() options: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Dial h: Allow the callee to hang up by dialing * H: Allow the caller to hang up by dialing * Perhaps it is some other system inline with yours that has this feature enabled. I certainly found this to be the case with a version of a2billing which included Hh in the dial parameters by default. It is quite possible that the same system is swallowing your other DTMF presses too. Regards, Steve On 4/4/07, Mike Diehl [EMAIL PROTECTED] wrote: There wasn't a setting, but I set it to rfc2833. On Wednesday 04 April 2007 12:49, Noah Miller wrote: Hi Mike - Well, when I restart the cli as requested below and go the addition steps of setting verbose to 25 and turning sip debug on for the phone in test, I don't get ANYTHING on the console. Sounds like it's a phone issue after all, right? I've got the same symptoms for BOTH the Sipura 2002 and a Polycom 501. Any ideas where to start? What is your dtmfmode set to in sip.conf (e.g. inband, rfc2833)? - Noah Thanx, Mike Diehl. On Thursday 29 March 2007 11:52, Doug wrote: At 18:23 3/28/2007, Mike Diehl wrote: Actually, it turns out that sometimes I can't get ANY DTMF to work. I can call a local phone number and log into my voicemail system at work. But my wife is unable to dial a toll free number and use their IVR. Hope this helps. What does your log read? asterisk -rvvv On Wednesday 28 March 2007 16:58, Mike Diehl wrote: Hi all, I've trying to fix a problem. If I'm in a call and I press the * key, the call goes silent but doesn't hang up. I need to be able to send the * key for various IVR's that I interact with. Since I thought this was related to the features.conf file, you can view it at: http://www.diehlnet.com/features.conf Any ideas are welcome. TIA, -- Mike Diehl ___ --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 601 message waiting indicator
Hi, I'm used to the Polycom 501, with a big message light indicator that flashes when I have a message waiting. As far as I can see when looking at pictures, the 601 (and 650) do not have this indicator light (nor does the 550 for that matter). How does it show the user that it has a message waiting? Regards, Mike ___ --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 601 message waiting indicator
It has one, you just can't see it as easily in photos. It is to the right top corner of the display, top edge of the phone. -- Chris Mason (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 UK 44.207.183.0271 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] Configuration assistance needed.
I have a fairly large system to configure. I was hoping to find someone locally to employ for this project but remote configuration is considerable. Pleas let me know if you are interested and have the time. ___ --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] What is this error message? (check_auth: stale nonce received from ...)
I`ve been noticing alot of those messages in the CLI lately: Apr 5 11:18:02 NOTICE[25593]: chan_sip.c:6444 check_auth: stale nonce received from 'sip:[EMAIL PROTECTED] I haven't changed my configuration in ages. What could be the cause of this suddent appearance? Mike ___ --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] remote SIP, no audio, or one way audio.
Steve Totaro [EMAIL PROTECTED] Wrote: 4/4/2007 8:44 PM: Joe Acquisto wrote: Attempts to do SIP thru firewall (IPCop) are unsuccessful. using x-lite softphones, for eval/testing. They do get registered, and can call each other, but mostly get no audio, sometimes one way audio. Suggestions/fixes? joe a. Is there NAT on both sides? Are you using qualify? Paint a clearer picture. Sorry, I missed your reply, till now. --switch | | |phones | |-asterisk box |---IPcop|---internet-|-home/remote-office|sip phone |-ditto Hope that is intelligible. joe a ___ --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] No response on extensions: TDM842
Hi all I recently bought a TDM842, basically a TDM800P card with 4 FXS, 2 FXO's. Installed the same driver as for the TDM24xx cards, as the manual says. Running [EMAIL PROTECTED] 2.8, did yum update, now on kernel 2.6.9-42.0.10.ELsmp Card config: Channel 1-4: FXS, Channel 5,6 FXO From dmesg: After resetting the modules... Port 1: Installed -- AUTO FXS/DPO Port 2: Installed -- AUTO FXS/DPO Port 3: Installed -- AUTO FXS/DPO Port 4: Installed -- AUTO FXS/DPO Port 5: Installed -- AUTO FXO (FCC mode) Port 6: Installed -- AUTO FXO (FCC mode) Port 7: Not installed Port 8: Not installed I have 3 normal analogue phones on channels 1,2,3 (plugs marked with 1,2,3 on the card) Created extentions via FreePBX for ZAP\1, ZAP\2, ZAP\3, dialing 1000, 2000, 3000 to reach respective extensions. Created trunk g0 and outgoing route to accees it if dialing 0, all via FreePBX Problem is, I can't dial between any of my extensions. I get a slight noise on the line, indicating to me that the channels are active on the card, but no matter what I do, I can't get any response. I can't dial to an outside number either. Basically, nothing works! Please assist, bearing in mind that this is my first time setting up Asterisk, so don't assume anything! Where can I start looking? Thanx in advance. Charl Papenfus ___ --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 601 message waiting indicator
Thanks David and Chris, appreciate the response Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave Fullerton Sent: Thursday, April 05, 2007 11:40 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom 601 message waiting indicator Mike wrote: Hi, I'm used to the Polycom 501, with a big message light indicator that flashes when I have a message waiting. As far as I can see when looking at pictures, the 601 (and 650) do not have this indicator light (nor does the 550 for that matter). How does it show the user that it has a message waiting? Regards, Mike It's there. There is a vertical groove on each side of the display and the indicator is at the top of the right groove. Check page 4 of the Soundpoint IP 601 Quick Start Guide for a nice diagram: http://www.polycom.com/common/pw_item_show_doc/1,1276,4886,00.pdf -Dave ___ --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 ___ --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 repair
Jesse, I would just add that if he has a 600 it's worth getting the 601. The 600 was limited in memory and had some other disadvantages. James I would hang on to the phone for testing or something else. Even though there may be no screen, you can program it via the web interface and it can be used for something. Dovid - Original Message - From: Jessee J Holmes To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Thursday, April 05, 2007 5:05 PM Subject: Re: [asterisk-users] polycom repair James Andrewartha, We send these into Polycom for repair on occasion. It's will cost roughly $200~$300 direct with Polycom and will take 8~10 weeks for repair. Obviously, most people opt not to do a repair and just buy a new phone instead. The only time it may make sense to repair an out of warranty phone would be if you have some sentimental attachment to your current phone or more realistically, you're not certain the newer phones will work with your PBX or SP. There may be other reasons, but I can't think of those right now. Keep in mind, IP 600's aren't available any longer from Polycom. The new replacement is the IP 601 phone or the IP 650. Jessee Holmes Atacomm / Ataractic Corporation www.atacomm.com V: 1-877-700-VOIP [EMAIL PROTECTED] Looking for voice over IP products? Visit our VoIP store at http://voipstore.atacomm.com/ On Apr 4, 2007, at 11:36 PM, James Andrewartha wrote: Hi all, Has anyone had any experience getting Polycom phones repaired? The screen on one of our IP600s got smashed, and I'm wondering if it's worth the effort to get it repaired, or if it'd just be cheaper to buy a new phone. Thanks, -- James Andrewartha Systems Administrator Data Analysis Australia Pty Ltd ___ --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 -- ___ --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 ___ --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] Configuration assistance needed.
What is locally ? Where are you located ? - Original Message - From: Tim King [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Thursday, April 05, 2007 6:15 PM Subject: [asterisk-users] Configuration assistance needed. I have a fairly large system to configure. I was hoping to find someone locally to employ for this project but remote configuration is considerable. Pleas let me know if you are interested and have the time. ___ --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 ___ --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] Best Hardphone (Subjective?)
Bill Hackensack wrote: On 4/2/07, *Corporate IT Solutions - Michael Dunne* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: So subjectively what would be the best Hardphone for a small/medium business with multiple line support, BLF, etc. Does _anyone_ read the archives anymore? This is like a weekly question or something. Oh look! Bill is back! -S ___ --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] IAX Trunk Failover
tried x+102 ?
On 4/5/07, Brent [EMAIL PROTECTED] wrote:
I'm trying to get an IAX trunk to failover to a local trunk it the trunk
is down.
This is what I've been working on:
[macro-forward1];
exten = s,1,Dial(IAX2/192.168.1.1/${ARG1},20)
exten = s,2,Goto(call-${DIALSTATUS},1)
exten = s-CONGESTION,1,Dial(LOCAL/${ARG2},20)
exten = s-CHANUNAVAIL,1,Dial(LOCAL/${ARG2},20
;end macro-forward1
exten = 6222626,1,Macro(forward1,6222626,6222627)
...in the debug, I never see dialstatus...the call just fails. Doesn't
ever try to dial the second extension.
Any ideas?
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--
Mike
Sales Manager
http://www.voicemeup.com
Making it happen
1.877.807.VOIP (8647)
1.514.312.7030
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Re: [asterisk-users] Best Hardphone (Subjective?)
Michael Graves wrote: I think that the Polycom phones are really good, but their lack of support for Asterisk is legendary. Though I'm no fan of some of Polycom's policies, this isn't strictly true anymore. They know which way the wind is blowing: http://forms.polycom.com/audio_files/techpartners.htm -Stephen- ___ --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] detecting a beep
at the moment, if our agents make a call and they get an answering machine they have to wait for the beep before leaving a message. I would like them to be able to transfer the call to an extension where an automated message can be left as soon as they know it is an A/M. However, how do I know how long to wait before playing back the message ? Is there a beep detector function ? Julian. ___ --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] Re: asterisk-users Digest, Vol 33, Issue 12
john beaman wrote: I too was curious about this, so I copied the text into Babel Fish, and this is the result: I miss of the 2/04/2007 to the 11/04/2007. I will answer your message as of my return. For any urgency, to contact Emmanuelle Parache Moga or Cédric Buzay. If this guy is really going to be out until November these messages will get rather tiresome... The rest of the world uses a sensible date format like DD/MM/ or /MM/DD :) (So it's April 11th, not November 4th) -Stephen- ___ --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 and Asterisk
Maybe the firmware uses GPL'd code? ;-) Just a theory, don't sue me Polycom! On 4/5/07, Stephen Bosch [EMAIL PROTECTED] wrote: Andrew Joakimsen wrote: Well I would wonder how Polycom even had any idea whom your vendor is. The vendor made a request for 2.1.0 on my behalf and let it slip that it was for one of my clients :) What is it about current firmware that makes them so paranoid? For pete's sake! If the argument is protecting IP, it's a lousy one. -Stephen- ___ --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 ___ --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] PRI DCHAN Errors
On 4/5/07, Rob Schall [EMAIL PROTECTED] wrote: Apr 4 12:13:03 WARNING[6670] chan_zap.c: No D-channels available! Using Primary channel 28 as D-channel anyway! Apr 4 12:13:05 WARNING[6660] channel.c: Avoided initial deadlock for '0x82b8430', 10 retries! Apr 4 12:13:05 WARNING[6660] channel.c: Avoided initial deadlock for '0x82d9920', 10 retries! Apr 4 12:13:05 WARNING[6660] channel.c: Avoided initial deadlock for '0x82369f0', 10 retries! Apr 4 12:13:05 WARNING[6660] channel.c: Avoided initial deadlock for '0x8242968', 10 retries! Apr 4 12:13:06 WARNING[6670] chan_zap.c: No D-channels available! Using Primary channel 28 as D-channel anyway! Apr 4 12:13:09 NOTICE[15466] app_dial.c: Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion) Apr 4 12:13:11 NOTICE[6670] chan_zap.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 2 I have read that some people believe this is a driver problem, which others believe something could be wrong with the PRI itself. I did a zttool and there are and haven't been any red alarms. I also read some people believe this issue can be caused by another card in the box taking too long to respond, such as an IDE card. We have 2 sangoma PRI cards and 1 sangoma FXO/FXS card. Any thoughts? FWIW I recently had some real troubles with something very similar, including the HDLC aborts, congestion, and terrible voice quality. Fortunately I caught the problem before the workweek so I could get it working. A little backstory: I had been running Linux 2.6.17-gentoo-r8 with Asterisk 1.2.14, Zaptel 1.2.12, libpri 1.2.4 and iaxmodem 0.2.0. As I was preparing to move all of my users over to Asterisk I wanted to upgrade before I moved them over so that I was on 1.4.x first. I also figured it wouldn't be a bad time to upgrade the kernel. (I know, one thing at a time) I upgraded to 2.6.19-gentoo-r5, Asterisk 1.4.2, Zaptel 1.4.1, libpri 1.4.0, and iaxmodem 0.2.1. After completing the upgrade the system would occasionally appear to hardware lock only to return to normal after 15-20 seconds. No process accounted for 100% cpu, however. It didn't do this all the time, maybe once every few hours. Of course this lockup would cause the PRIs on the system (quad port T1 card) to go a little nuts. VoIP calls would stutter terribly. I ended up in a rollback-reupgrade process that left me with kernel 2.6.17-gentoo-r8, the original kernel, but with the upgraded applications and libraries. I'm not convinced the problem was 2.6.19-gentoo-r5 as I also tried a 2.6.20 kernel and I have not seen anyone else asking about this problem but the evidence is pointing that way. During this process I thought for a moment I had lost my TE410P as it stopped talking to Asterisk but would talk to ztcfg/zttool/etc. Debug on the span showed Sending Set Asynchronous Balanced Mode Extended endlessly. I rolled back to my previous application setup and the server came up fine. So I rolled forward again with the apps without the kernel upgrade and it appears to work fine now. I'm running * on an IBM x306 server, 8836-1SU, specifically, with a TE410P. At some point I will attempt to upgrade the kernel again and see if that was the problem or I just had something screwed up. I don't know if that will help you but that was last weekend's hours of fun for me. :) ___ --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] Dialplan not reading MySQL table
Hello, I'm trying to use MySQL for Dialplans and have followed the Asterisk RealTime Extensions setup. The MySQL table is called extensions and I have entered two records.. ext 1000 and 2000. I also added switch = Realtime/[EMAIL PROTECTED] in extensions.conf and extensions = mysql,asterisk,extensions in extconfig.conf I do a *CLI dialplan reload but when I show the dialplan it has 0 extensions www2*CLI dialplan show test [ Context 'test' created by 'pbx_config' ] Alt. Switch ='Realtime/[EMAIL PROTECTED]' [pbx_config] -= 0 extensions (0 priorities) in 1 context. =- I don't get any errors in the debug or messages files. MySQL is working for both CDR and SIP without any problems. thanks Doug ___ --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] disabling authentication
From: Mark Price [EMAIL PROTECTED] Date: Wed, 4 Apr 2007 10:07:31 -0400 Is there a way to cause asterisk to accept all calls without any authentication? Mark Yes - not to set up a user/peer section in sip.conf. The context in [general] section will be used. Yuan Liu ___ --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] PRI DCHAN Errors
On 4/5/07, Rob Schall [EMAIL PROTECTED] wrote: Hey all, I had a user complaining of calls which were dropping mid-conversation. I looked into the time of one of the calls, and saw the following: Apr 4 12:13:03 WARNING[6670] chan_zap.c: No D-channels available! Using Primary channel 28 as D-channel anyway! Apr 4 12:13:05 WARNING[6660] channel.c: Avoided initial deadlock for '0x82b8430', 10 retries! Apr 4 12:13:05 WARNING[6660] channel.c: Avoided initial deadlock for '0x82d9920', 10 retries! Apr 4 12:13:05 WARNING[6660] channel.c: Avoided initial deadlock for '0x82369f0', 10 retries! Apr 4 12:13:05 WARNING[6660] channel.c: Avoided initial deadlock for '0x8242968', 10 retries! Apr 4 12:13:06 WARNING[6670] chan_zap.c: No D-channels available! Using Primary channel 28 as D-channel anyway! Apr 4 12:13:09 NOTICE[15466] app_dial.c: Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion) Apr 4 12:13:11 NOTICE[6670] chan_zap.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 2 I have read that some people believe this is a driver problem, which others believe something could be wrong with the PRI itself. I did a zttool and there are and haven't been any red alarms. I also read some people believe this issue can be caused by another card in the box taking too long to respond, such as an IDE card. We have 2 sangoma PRI cards and 1 sangoma FXO/FXS card. Any thoughts? 1) Check the connection between the demarc and your card. Is the cable the proper specification? 2) Have you updated zaptel or wanpipe recently? If you created a symlink to /usr/src/zaptel make sure its correct. If its not correct and reinstall. 3) Do you have the correct FE_LCODE, FE_FRAME, TE_CLOCK and TDMV_DCHAN in your wanpipe1.conf? Do you have the correct signalling in zapata.conf. Do you have the correct span definition and fcshdlc=24, fcshdlc=48 (unless those analog ports are defined first!) in zaptel.conf 4) Use wanpipemon -g instead of zttool. See if you are getting any errors 5) Contact your telco have them check the line Take a look at this as well: http://www.voip-info.org/wiki/view/Asterisk+PCI+bus+Troubleshooting I found it rather interesting. ___ --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 and Asterisk
Stephen Bosch wrote: Andrew Joakimsen wrote: Well I would wonder how Polycom even had any idea whom your vendor is. The vendor made a request for 2.1.0 on my behalf and let it slip that it was for one of my clients :) What is it about current firmware that makes them so paranoid? For pete's sake! If the argument is protecting IP, it's a lousy one. -Stephen- IP is just part of the smoke and mirrors. Another part is quality of the service experience. The real issue is control of the sales channel. This allows them to keep prices high for Polycom and their dealers, same as Nortel, Avaya et al used to do. regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ --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] Re: asterisk-users Digest, Vol 33, Issue 12
Since when is Canada part of the rest of the world? I thought it was a US National Park? ;-) On 4/5/07, Stephen Bosch [EMAIL PROTECTED] wrote: john beaman wrote: I too was curious about this, so I copied the text into Babel Fish, and this is the result: I miss of the 2/04/2007 to the 11/04/2007. I will answer your message as of my return. For any urgency, to contact Emmanuelle Parache Moga or Cédric Buzay. If this guy is really going to be out until November these messages will get rather tiresome... The rest of the world uses a sensible date format like DD/MM/ or /MM/DD :) (So it's April 11th, not November 4th) -Stephen- ___ --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 ___ --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] Asterisknow or Trixbox?
I am sure its been discussed before but I couldn't find it in my searches.. Looking to replace my Asterisk box (Ver 1.0 still I think) and really like the idea of an easy to use gui to manage it.. I see the contenders appear to be Asterisknow and Trixbox.. Has anyone player with both who can give me the rundown of the basic pros and cons to either.. Which will best suit my setup??.. Currently I have 3 inbound numbers that come in over IAX plus one in/out analogue line to an X100P.. I have a DECT phone extension connecting to a TDM400P.. There are three local SIP extension and one in a remote office.. Inbound numbers are routed to a recorded message specific for that number and then set to ring on multiple phones (different combinations depending on the inbound number) and if not answered routed to a VM box.. Caller ID is changed on the inbound calls so that the phone displays will show which number was called rather than the calling number and so we can answer with the correct greeting.. The analogue inbound calls simply ring on all the phones.. All outbound calls are via the analogue line and use a carrier pre selection prefix for cheap outbound calls.. So which would you suggest I use? Both are in beat versions at the moment.. Which is likely to so stable first? Thanks ___ --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] IAX2 threads
I've been running an asterisk box that is routing an average of 9 simultaneous calls between a PRI and five asterisk boxes via IAX. After having it run for about five hours I had a spate of these error messages: Out of idle IAX2 threads for I/O, pausing! And then they went away. The only references I can find on google for this error message are in the source. What does this mean and if it's a problem what should I be looking for to diagnose the cause? TIA Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] ___ --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] Re: asterisk-users Digest, Vol 33, Issue 22
Je suis absent du 2/04/2007 au 11/04/2007. Je répondrai à votre message dès mon retour. Pour toute urgence, contacter Emmanuelle Parache Moga ou Cédric Buzay. ___ --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] Queue call distribution
I have noticed that asterisk will only try one interface per queue at a time. Is there any way get get it to dial say three at a time and connect the first one that it reaches. ___ --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] PRI DCHAN Errors
Andrew Joakimsen wrote: On 4/5/07, Rob Schall [EMAIL PROTECTED] wrote: Hey all, I had a user complaining of calls which were dropping mid-conversation. I looked into the time of one of the calls, and saw the following: Apr 4 12:13:03 WARNING[6670] chan_zap.c: No D-channels available! Using Primary channel 28 as D-channel anyway! Apr 4 12:13:05 WARNING[6660] channel.c: Avoided initial deadlock for '0x82b8430', 10 retries! Apr 4 12:13:05 WARNING[6660] channel.c: Avoided initial deadlock for '0x82d9920', 10 retries! Apr 4 12:13:05 WARNING[6660] channel.c: Avoided initial deadlock for '0x82369f0', 10 retries! Apr 4 12:13:05 WARNING[6660] channel.c: Avoided initial deadlock for '0x8242968', 10 retries! Apr 4 12:13:06 WARNING[6670] chan_zap.c: No D-channels available! Using Primary channel 28 as D-channel anyway! Apr 4 12:13:09 NOTICE[15466] app_dial.c: Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion) Apr 4 12:13:11 NOTICE[6670] chan_zap.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 2 I have read that some people believe this is a driver problem, which others believe something could be wrong with the PRI itself. I did a zttool and there are and haven't been any red alarms. I also read some people believe this issue can be caused by another card in the box taking too long to respond, such as an IDE card. We have 2 sangoma PRI cards and 1 sangoma FXO/FXS card. Any thoughts? 1) Check the connection between the demarc and your card. Is the cable the proper specification? Yes, the connection seems solid and the cable is alright. 2) Have you updated zaptel or wanpipe recently? If you created a symlink to /usr/src/zaptel make sure its correct. If its not correct and reinstall. There have not been any updates to zaptel or wanpipe in awhile 3) Do you have the correct FE_LCODE, FE_FRAME, TE_CLOCK and TDMV_DCHAN in your wanpipe1.conf? Do you have the correct signalling in zapata.conf. Do you have the correct span definition and fcshdlc=24, fcshdlc=48 (unless those analog ports are defined first!) in zaptel.conf Here is my wanpipe1.conf [devices] wanpipe1 = WAN_AFT_TE1, Comment [interfaces] w1g1 = wanpipe1, , TDM_VOICE, Comment [wanpipe1] CARD_TYPE = AFT S514CPU = A CommPort= PRI AUTO_PCISLOT= NO PCISLOT = 1 PCIBUS = 1 FE_MEDIA= T1 FE_LCODE= B8ZS FE_FRAME= ESF FE_LINE = 1 TE_CLOCK= NORMAL TE_REF_CLOCK= 0 TE_HIGHIMPEDANCE= NO LBO = 0DB FE_TXTRISTATE = NO MTU = 1500 UDPPORT = 9000 TTL = 255 IGNORE_FRONT_END = NO TDMV_SPAN = 1 TDMV_DCHAN = 0 [w1g1] ACTIVE_CH = ALL TDMV_ECHO_OFF = NO TDMV_HWEC = YES 4) Use wanpipemon -g instead of zttool. See if you are getting any errors Everything is connected and reading okay. 5) Contact your telco have them check the line In order for our provider to do this, they have to shut down the pri. This could be done at night, but wanted to make sure I had everything else in place. Take a look at this as well: http://www.voip-info.org/wiki/view/Asterisk+PCI+bus+Troubleshooting I found it rather interesting. ___ --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 ___ --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] Analog phones, dial out
I have a system with a TDM400p 2FXO, 2FXS. Analog phones work fine, on incoming calls, ring, answer, talk, hangup. However, not so good dialing out. Pickup handset, get dail tone. Cli shows Starting simple switch on Zap/1-1. Press a key and get Hungup Zap/1-1 and get the doot-doot-doot sound. Well, something like that, anyway. It may take a large 2x4, today. joe a. ___ --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] Queue call distribution
Jordan Novak wrote: I have noticed that asterisk will only try one interface per queue at a time. Is there any way get get it to dial say three at a time and connect the first one that it reaches. Do you have autofill=yes in queues.conf? Regards, Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --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] Queue call distribution
If you set the queue strategy to ringall it should ring all the interfaces you have set up in that queue. Just make sure you have member = SIP/EXT setup. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jordan Novak Sent: Thursday, April 05, 2007 4:06 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Queue call distribution I have noticed that asterisk will only try one interface per queue at a time. Is there any way get get it to dial say three at a time and connect the first one that it reaches. ___ --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] Asterisknow or Trixbox?
On Thu, 5 Apr 2007, WipeOut wrote: I am sure its been discussed before but I couldn't find it in my searches.. Looking to replace my Asterisk box (Ver 1.0 still I think) and really like the idea of an easy to use gui to manage it.. I see the contenders appear to be Asterisknow and Trixbox.. How about no GUI? Has anyone player with both who can give me the rundown of the basic pros and cons to either.. Which will best suit my setup??.. Currently I have 3 inbound numbers that come in over IAX plus one in/out analogue line to an X100P.. I have a DECT phone extension connecting to a TDM400P.. There are three local SIP extension and one in a remote office.. Inbound numbers are routed to a recorded message specific for that number and then set to ring on multiple phones (different combinations depending on the inbound number) and if not answered routed to a VM box.. Caller ID is changed on the inbound calls so that the phone displays will show which number was called rather than the calling number and so we can answer with the correct greeting.. The analogue inbound calls simply ring on all the phones.. All outbound calls are via the analogue line and use a carrier pre selection prefix for cheap outbound calls.. So which would you suggest I use? Both are in beat versions at the moment.. Which is likely to so stable first? It doesn't look like your setup will change once it's going, so why burden the system with something that's quite general purpose? Take a look at the dialplans (extensions.conf), etc. that your current setup has and just build on that directly withou a GUI... Gordon ___ --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] Asterisknow or Trixbox?
I also am curious to hear feedback on either asterisknow and/or trixbox. I am looking to install a few boxes in locations that people need to be able to add extensions (queues, etc), but might not be avid linux users. I could write my own gui, but why bother. A nice easy installation and extension/queue/voicemail management system would be nice. Rob Gordon Henderson wrote: On Thu, 5 Apr 2007, WipeOut wrote: I am sure its been discussed before but I couldn't find it in my searches.. Looking to replace my Asterisk box (Ver 1.0 still I think) and really like the idea of an easy to use gui to manage it.. I see the contenders appear to be Asterisknow and Trixbox.. How about no GUI? Has anyone player with both who can give me the rundown of the basic pros and cons to either.. Which will best suit my setup??.. Currently I have 3 inbound numbers that come in over IAX plus one in/out analogue line to an X100P.. I have a DECT phone extension connecting to a TDM400P.. There are three local SIP extension and one in a remote office.. Inbound numbers are routed to a recorded message specific for that number and then set to ring on multiple phones (different combinations depending on the inbound number) and if not answered routed to a VM box.. Caller ID is changed on the inbound calls so that the phone displays will show which number was called rather than the calling number and so we can answer with the correct greeting.. The analogue inbound calls simply ring on all the phones.. All outbound calls are via the analogue line and use a carrier pre selection prefix for cheap outbound calls.. So which would you suggest I use? Both are in beat versions at the moment.. Which is likely to so stable first? It doesn't look like your setup will change once it's going, so why burden the system with something that's quite general purpose? Take a look at the dialplans (extensions.conf), etc. that your current setup has and just build on that directly withou a GUI... Gordon ___ --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 ___ --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] Re: ZAP device reference in Zaptel 1.4
Hi, On Wed, Apr 04, 2007 at 04:48:08PM +0300, Tzafrir Cohen wrote: On Wed, Apr 04, 2007 at 10:11:01AM +0300, Tzafrir Cohen wrote: On Wed, Apr 04, 2007 at 04:59:41PM +1000, Devraj Mukherjee wrote: monk*CLI zap show channels No such command 'zap show' (type 'help' for help) Does that mean I dont have ZAP support in Asterisk? Maybe. ls -l /usr/lib/asterisk/modules/chan_zap.so I also repeat my second question: What is the contents of /etc/asterisk/zapata.conf ? Follow-up: The issue seems to be an issue with the atrpms package: http://bugzilla.atrpms.net/show_bug.cgi?id=1165 Asterisk 1.4.2 is missing chan_zap.so Thanks to Tzafrir, Mark, Eric and other the bug was identified and sqashed (it was a wrong build order that had asterisk 1.4.2 build against zaptel 1.4.0 instead of 1.4.1). Updated packages are on their way (1.4.2-37) to the master repo, so please yum update to get chan_zap support back. -- Axel.Thimm at ATrpms.net pgpjUXqFq57Ke.pgp Description: PGP signature ___ --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] Analog phones, dial out
Paste here the rules of your extensions.conf for outgoing calls. Sds, Gustavo From: Joe Acquisto [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] Analog phones, dial out Date: Thu, 05 Apr 2007 16:28:44 -0400 I have a system with a TDM400p 2FXO, 2FXS. Analog phones work fine, on incoming calls, ring, answer, talk, hangup. However, not so good dialing out. Pickup handset, get dail tone. Cli shows Starting simple switch on Zap/1-1. Press a key and get Hungup Zap/1-1 and get the doot-doot-doot sound. Well, something like that, anyway. It may take a large 2x4, today. joe a. ___ --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 _ Verificador de Segurança do Windows Live OneCare: combata já vírus e outras ameaças! http://onecare.live.com/site/pt-br/default.htm ___ --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] BeroNet HFC-4S card is now detected as only 2 ports
Hello list, After upgrading from BRIstuff 1y-b to 1y-e my ISDN card is suddently detected as 2 ports instead of 4. I still load the driver as modprobe qozap ports=12 as I've always done. But now it only sees 2 ports. Output of lspci -vvv -- cut 02:01.0 ISDN controller: Cologne Chip Designs GmbH ISDN network Controller [HFC-4S] (rev 01) Subsystem: Cologne Chip Designs GmbH Unknown device b560 Control: I/O+ Mem+ BusMaster- SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=medium TAbort- TAbort- MAbort- SERR- PERR- Interrupt: pin A routed to IRQ 22 Region 0: I/O ports at ddb8 [size=8] Region 1: Memory at fcefa000 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 Flags: PMEClk- DSI+ D1+ D2+ AuxCurrent=0mA PME(D0+,D1+,D2+,D3hot+,D3cold-) Status: D0 PME-Enable- DSel=0 DScale=0 PME- -- cut Anyone has any ideas? -- Best regards, Henrik Woffinden ___ --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] Open Source VoIP client (on a webpage)
I need to decide on the best way to add a voip SIP or IAX client to a website. I'm thinking that I'd like it to be inline, like an aplet, on the page. I've got some asterisk servers running to connect up to, so the real challenge is finding an easily integrated open source client. Any suggestions from those who know? Jason ___ --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] PRI DCHAN Errors
On 4/5/07, Rob Schall [EMAIL PROTECTED] wrote: Yes, the connection seems solid and the cable is alright. That doesn't mean the cable is the proper specification. Most people use category 5 unshielded twisted pair which technically is not the correct cable. While this is probably not the cause of your problems when troubleshooting is wise not to assume anything. There have not been any updates to zaptel or wanpipe in awhile Zaptel 1.2.16 released 21 March 2007 and the latest WANPIPE is released only since 31 January, so if your install is at least 1 month old there have been changes to the software. If this issue has cropped up recently and you haven't touched your zaptel or wanpipe then DO NOT UPGRADE THEM as part of your troubleshooting 3) Do you have the correct FE_LCODE, FE_FRAME, TE_CLOCK and TDMV_DCHAN in your wanpipe1.conf? Do you have the correct signalling in zapata.conf. Do you have the correct span definition and fcshdlc=24, fcshdlc=48 (unless those analog ports are defined first!) in zaptel.conf Here is my wanpipe1.conf TDMV_DCHAN = 0 That doesn't mean that the settings are correct, however chances are your PRI is indeed B8ZS, ESF, most are as well. And it still is possible that some settings in zaptel.conf and zapata.conf are incorrect. However you MIGHT want to try to set TDMV_DCHAN to 24 which will enable hardware Dchannel support on the Sangoma card, it could be you are having interrupt, latency or congestion problems on your PCI bus that cause the issues with the DChannel to to Software (Zaptel) D channel handling. 4) Use wanpipemon -g instead of zttool. See if you are getting any errors Everything is connected and reading okay. So all the error counts are 0? 5) Contact your telco have them check the line In order for our provider to do this, they have to shut down the pri. This could be done at night, but wanted to make sure I had everything else in place. They should be able to do some testing between the smartjack and the network without too many problems. Do you know if the T1 is run over fiber or if its HDSL? I have never seen a problem with a fiber-fed T1 either it works or it doesnt. HDSL however is another story it can be just as bad as ADSL if the telco does a poor job to condition the line. If you are really reluctant to do any testing with the telco then perhaps you can try to contact Sangoma. I've never heard anything bad of their tech support. ___ --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] Call dies when I press *
Man, I really thought you had nailed it. So I defined the variable that holds those paramters to the empty string and tried again. From my console log: Dial(SIP/line_3-b7701f30, IAX2/Uxxx/18003310500|90|) in new stack I still have the same symptoms. However, if I dial 5058457900, I'm able to navigate just fine. Any other ideas would be more than welcome! TIA, Mike. On Thursday 05 April 2007 08:47, Steve Davies wrote: Is it related to Dial() options: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Dial h: Allow the callee to hang up by dialing * H: Allow the caller to hang up by dialing * Perhaps it is some other system inline with yours that has this feature enabled. I certainly found this to be the case with a version of a2billing which included Hh in the dial parameters by default. It is quite possible that the same system is swallowing your other DTMF presses too. Regards, Steve On 4/4/07, Mike Diehl [EMAIL PROTECTED] wrote: There wasn't a setting, but I set it to rfc2833. On Wednesday 04 April 2007 12:49, Noah Miller wrote: Hi Mike - Well, when I restart the cli as requested below and go the addition steps of setting verbose to 25 and turning sip debug on for the phone in test, I don't get ANYTHING on the console. Sounds like it's a phone issue after all, right? I've got the same symptoms for BOTH the Sipura 2002 and a Polycom 501. Any ideas where to start? What is your dtmfmode set to in sip.conf (e.g. inband, rfc2833)? - Noah Thanx, Mike Diehl. On Thursday 29 March 2007 11:52, Doug wrote: At 18:23 3/28/2007, Mike Diehl wrote: Actually, it turns out that sometimes I can't get ANY DTMF to work. I can call a local phone number and log into my voicemail system at work. But my wife is unable to dial a toll free number and use their IVR. Hope this helps. What does your log read? asterisk -rvvv On Wednesday 28 March 2007 16:58, Mike Diehl wrote: Hi all, I've trying to fix a problem. If I'm in a call and I press the * key, the call goes silent but doesn't hang up. I need to be able to send the * key for various IVR's that I interact with. Since I thought this was related to the features.conf file, you can view it at: http://www.diehlnet.com/features.conf Any ideas are welcome. TIA, -- Mike Diehl ___ --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 -- Mike Diehl ___ --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] Asterisknow or Trixbox?
I know you will receive many replies to your request. We have selected Thirdlane PBX Manager (www.thirdlane.com) to manage our Asterisk installations. The key for us was a management system that was modest in cost and allowed us to easily provide customizations when necessary, yet allow the system to be maintained by a non linux sysadmin or office manager. The system is a Webmin module and allows the administrator to manage virtually all aspects of Asterisk without needing to go to linux CLI. This includes phone provisioning and a user web portal so individual users can manage their own voicemail and simple call routing (thus reducing sysadmin burden and increasing user acceptance). Feel free to contact me offline if I can be of any assistance. Regards, Jim [EMAIL PROTECTED] - Original Message - From: Rob Schall [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, April 5, 2007 2:23:44 PM (GMT-0800) America/Los_Angeles Subject: Re: [asterisk-users] Asterisknow or Trixbox? I also am curious to hear feedback on either asterisknow and/or trixbox. I am looking to install a few boxes in locations that people need to be able to add extensions (queues, etc), but might not be avid linux users. I could write my own gui, but why bother. A nice easy installation and extension/queue/voicemail management system would be nice. Rob Gordon Henderson wrote: On Thu, 5 Apr 2007, WipeOut wrote: I am sure its been discussed before but I couldn't find it in my searches.. Looking to replace my Asterisk box (Ver 1.0 still I think) and really like the idea of an easy to use gui to manage it.. I see the contenders appear to be Asterisknow and Trixbox.. How about no GUI? Has anyone player with both who can give me the rundown of the basic pros and cons to either.. Which will best suit my setup??.. Currently I have 3 inbound numbers that come in over IAX plus one in/out analogue line to an X100P.. I have a DECT phone extension connecting to a TDM400P.. There are three local SIP extension and one in a remote office.. Inbound numbers are routed to a recorded message specific for that number and then set to ring on multiple phones (different combinations depending on the inbound number) and if not answered routed to a VM box.. Caller ID is changed on the inbound calls so that the phone displays will show which number was called rather than the calling number and so we can answer with the correct greeting.. The analogue inbound calls simply ring on all the phones.. All outbound calls are via the analogue line and use a carrier pre selection prefix for cheap outbound calls.. So which would you suggest I use? Both are in beat versions at the moment.. Which is likely to so stable first? It doesn't look like your setup will change once it's going, so why burden the system with something that's quite general purpose? Take a look at the dialplans (extensions.conf), etc. that your current setup has and just build on that directly withou a GUI... Gordon ___ --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 ___ --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 ___ --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] IAX Trunk Failover
You have call-${DIALSTATUS} and s-CONGESTION.
It might not be CONGESTION. Do a Noop(${DIALSTATUS}) you should get
something.
Justin
On 4/5/07, Mike Lynchfield [EMAIL PROTECTED] wrote:
tried x+102 ?
On 4/5/07, Brent [EMAIL PROTECTED] wrote:
I'm trying to get an IAX trunk to failover to a local trunk it the
trunk is down.
This is what I've been working on:
[macro-forward1];
exten = s,1,Dial(IAX2/192.168.1.1/${ARG1},20)
exten = s,2,Goto(call-${DIALSTATUS},1)
exten = s-CONGESTION,1,Dial(LOCAL/${ARG2},20)
exten = s-CHANUNAVAIL,1,Dial(LOCAL/${ARG2},20
;end macro-forward1
exten = 6222626,1,Macro(forward1,6222626,6222627)
...in the debug, I never see dialstatus...the call just fails. Doesn't
ever try to dial the second extension.
Any ideas?
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--
Mike
Sales Manager
http://www.voicemeup.com
Making it happen
1.877.807.VOIP (8647)
1.514.312.7030
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Re: [asterisk-users] Open Source VoIP client (on a webpage)
Jason Wolfe wrote: I need to decide on the best way to add a voip SIP or IAX client to a website. I'm thinking that I'd like it to be inline, like an aplet, on the page. I've got some asterisk servers running to connect up to, so the real challenge is finding an easily integrated open source client. Any suggestions from those who know? Jason Look for a java applet, like JAIN Applet-Phone (http://snad.ncsl.nist.gov/proj/iptel/). There are at least a handful there, relatively easy to find via google. ___ --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] HPEC audio clipping
I have recently moved an asterisk system to a new location. This location is experiencing terrible echo. I installed the HPEC from Digium but that has caused a new problem. When HPEC is enabled and more that 16 taps are used, the audio from the outside caller gets clipped. Instead of hearing: Hello, my name is Mike one hears He o, m ameike If the taps are set to less than 16, there is no clipping, but there is significant echo. I don't know if this is relevant, but asterisk does report No Zaptel transcoder support on startup. I am at my wits end; any advice would be greatly appreciated. My setup follows: Hardware, AMD Athlon(tm) 64 Processor 3000+, 512MB ram, 2 TDM400p with a total of 5 fxo channels, snom 320 phones OS: gentoo 2006.1 amd64 Software: Asterisk 1.2.17 and Zaptel 1.2.16 (fxotune does not seem to work in this configuration) Also tried Asterisk 1.4.2 and Zaptel 1.4.1 (fxotune does work, but doesn't seem to help in this configuration) ___ --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
