Re: [Asterisk-Users] Cameron Palmer / voiceglo
At 23:40 30/01/04 -0700, you wrote: I found a message in the archives from Cameron Palmer on 23 Dec regarding his voiceglo SIP configuration. Unfortunately (for me), the archive has his email address removed. So, Cameron -- or anybody else using voiceglo with their * box -- please reply to me so that I can get your email address and ask you a question about your setup. Thanks, Greg ___ 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 Cameron's email address is Cameron Palmer [EMAIL PROTECTED] Peter Brown ___ 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] smtp question
Correct me if I'm wrong here but when a message bounces and the mailer/mta generates a bounce message shouldn't the from field have in it instead of an email addres (ie. [EMAIL PROTECTED]). The list was nailed with over 13,000 bounce messages(and they keep coming) from ONE list subscriber and I wasn't sure if their smtp server wasn't doing what it should. I finally had to just block their domain totally at postini to stop the flood of messages. I think this was part of the first slowdown and why the list slowed down again today. bkw ___ 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] P2P RTP without SIP re-invites
Let's go through how SIP works in Asterisk compared with a SIP Proxy. Remember that Asterisk is not designed to be a SIP Proxy, it's designed to be a Multi-VOIP and PSTN PBX, a quite complicated task. (I'm not going into all details (ACK, TRYING, RINGING etc)) We have two SIP users, Alice and Bob. Alice calls BOB, both connected to Asterisk: * Alice's UA sends an INVITE to [EMAIL PROTECTED] * Asterisk checks if bob is a valid user reachable within the context allowed by Alice's account * Asterisk answers the SIP call from Alice * Asterisk initiates another SIP call to Bob's UA with a NEW Invite * When Bob answers, Asterisk bridges the streams, performing codec conversion if necessary In this scenario, we now have two different SIP dialogues (two separate SIP calls) If both Alice and Bob are connected without NAT, have the same codec support and have canreinvite=yes * Asterisk send (re-)INVITEs to both, trying to get the RTP stream transferred so it goes directly from Alice to Bob Not all UAs support a re-INVITE and in public scenarios, a lot of UAs have broken NAT support so the RTP media stream stays with Asterisk. The benefit of this is that Asterisk acting as a user agent server (Alice) and client (bob) can send early media to Alice, connect to voicemail or another extension than Bob if Bob had issued a forward - maybe a H.323 connection or PSTN connection somewhere. With a SIP proxy we have the following scenario: * Alice's UA sends an INVITE to [EMAIL PROTECTED] * The proxy responsible for thte domain receives this and looks up bob in a user location or alias table * The proxy *FORWARDS* the same INVITE to [EMAIL PROTECTED], maybe several different locations * When Bob answers somewhere, the proxy cancels the call to the non-answering locations and forwards the OK to Alice * Alice ACKs the OK to bob and the call is UP In this scenario, there's only one SIP dialogue, between Alice and Bob with the proxy in the middle of signalling, but acting as a proxy and not as a user agent (the proxy can't and should not answer or originate calls). --- So, back to the original question, in a large installation (many users) - how do you off-load Asterisk? There's no single truth here, but here's my opinion: * If you are all on the same internal network, make sure the SIP phones support re-invites and use that. * If you have users all over the Internet, use a SIP proxy as a front-end to Asterisk You will still be forced to handle a lot of RTP streams (because of NAT), but can distribute that over a SIP-proxy network with SRV records, DNS round-robin techniques or forcing the users to register with different proxies. There's been a couple of suggestions that we should make Asterisk a good SIP proxy. If you spend some time learning to understand Asterisk's architecture, you'll also understand that this would not really work. I'm not saying the SIP channel can't be improved, I'm just saying that it has to work with the rest of Asterisk's architecture. I might be totally wrong, but my gut feeling is that Asterisk in combination with a separate SIP proxy is a very powerful solution. Clustering Asterisk servers somehow is also a good approach, but not here yet for SIP. /O ___ 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] newbie thinclient env
i have a lot of questions, maybe you've got some answers. I've got the following setup : 1 central server ( linux ) , 17 servers ( LTSP ) on different locations, where each server has a number of thinclients connected to it ( max 6 ) The central server / 17 LTSP servers are on the internet and share 1 VPN ( freeswan ). It would be nice if the 17 LTSP servers could use asterix on one way or another ( the central server is on a distant location , serves as a VPN gateway ). - asterix on the LTSP servers ? - can i connect a phone to the thinclients via USB ( some thinclients have no PCI extension possibilities ) - 17 LTSP servers have dynamic ip addresses ( here in belgium, that means that every 1 or 2 days, the ip address changes ). Can i use dyndns for asterix ? - does asterix imply a serious load on the server, say for 5 telephones - where do i buy these dev kits ( belgium ? ) - a lot more questions, jef peeraer ___ 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] Firefly and asterisk*
Hi Adam, I just got it to work ;)) I added an entry at the bottom of my iax.conf : register = *MY_FIREFLY_NUMBER*:[EMAIL PROTECTED] [firefly] type=friend host=firefly.virbiage.com context=incoming then, when a firefly user calls me, he is taken to incoming/s. I'm not sure if type=friend is right and if any other options should be set but IT WORKS!!! The only firefly related problem I'm still having it is that firefly erases the leadig 00 from every number in my (externel) contacts-list. bye and thanks - Original Message - From: Adam Hart [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, January 31, 2004 2:32 AM Subject: Re: [Asterisk-Users] Firefly and asterisk* - Original Message - From: FastJack [EMAIL PROTECTED] GREAT!!! Just got my asterisk* calling firefly users. Setup was really easy: snip Anyone knows how to receive calls on my asterisk*-box from the firefly-network? I'll fix this soon, then you should be able to connect to firefly network just like a normal iax2 connection. ___ 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] asterisk with big number of extentions.
Fran Boon wrote: Anton wrote: you can do it with a well setup cluster OK, so what success have people had with which clustering technologies? I'm more interested in resilience than performance. I would think that failover clustering would be far easier than a load sharing or processing cluster.. For lots of info on various clustering a HA systems take a look at http://www.linux-ha.org/ later.. ___ 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] determining legal VoIP service
Hi, -Original Message- Actually I believe this is one of the few things that can be done without worrying about the state(s) PUC coming down on your head. Since your users are in another country the state PUC cannot consider you providing a telephone service in their jurisdiction. Actually, this is precisely what matters. Your services can be considered to be within jurisdiction of your geographical area, or within the jurisdiction of your customers geographical area. Not both in most cases :-) I'm not sure which would apply for you - even within europe, standpoints differ... Florian ___ 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] newbie thinclient env
Hi! - asterix on the LTSP servers ? It is not advisable to run Asterisk on a server that runs X-Windows. You could give it a try on one machine, but I might very well turn out that you'll not like the resulting voice quality (choppy sound). - can i connect a phone to the thinclients via USB ( some thinclients have no PCI extension possibilities ) First thing is to check with LTSP if it supports local USB devices. And have you arranged already arranged sound support for the clients at all? Anyway, I'd rather go for a standard hardware IP phone - unless you are running a call center. - 17 LTSP servers have dynamic ip addresses ( here in belgium, that means that every 1 or 2 days, the ip address changes ). Can i use dyndns for asterix ? I wouldn't rely on that, dyndns can be slow and out-of-date. Instead register the dynamic Asterisk boxes with one or two static Asterisk boxes, which would basically mean that you'd end up with a star topolgy. - does asterix imply a serious load on the server, say for 5 telephones That depends on the codecs that you plan to use (which in turn depends on the phones that you use), as well as the number of concurrent calls. If transcoding is involved (=translation from one codec to another) then your CPU will be less bored. You should also consider QoS issues when running VoIP in a LTSP environment (for the record: Linux Terminx Server Project). Finally: Do you need your telephones to work even when the local LTSP server is down? Check the Wiki for codec details, bandwidth usage etc: http://www.voip-info.org/tiki-index.php?page=Asterisk Cheers, Philipp ___ 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] P2P RTP without SIP re-invites
Hi. If both Alice and Bob are connected without NAT, have the same codec support and have canreinvite=yes * Asterisk send (re-)INVITEs to both, trying to get the RTP stream transferred so it goes directly from Alice to Bob Not all UAs support a re-INVITE and in public scenarios, a lot of UAs have broken NAT support so the RTP media stream stays with Asterisk. a small correction: doesn't matter if Alice and Bob are nat'ed: if they're both nat'ed re-INVITEs are sent and RTP is transferred to go directly from Alice to Bob. Asterisk manages only the signalling on port 5060 (I'm using that environment, so it works :) ) But if only Alice OR Bob are nat'ed, the RTP is handled by * itself. Matteo. -- Brancaleoni Matteo [EMAIL PROTECTED] Espia - Emmegi Srl ___ 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] P2P RTP without SIP re-invites
Brancaleoni Matteo wrote: Hi. If both Alice and Bob are connected without NAT, have the same codec support and have canreinvite=yes * Asterisk send (re-)INVITEs to both, trying to get the RTP stream transferred so it goes directly from Alice to Bob Not all UAs support a re-INVITE and in public scenarios, a lot of UAs have broken NAT support so the RTP media stream stays with Asterisk. a small correction: doesn't matter if Alice and Bob are nat'ed: if they're both nat'ed re-INVITEs are sent and RTP is transferred to go directly from Alice to Bob. Asterisk manages only the signalling on port 5060 (I'm using that environment, so it works :) ) But if only Alice OR Bob are nat'ed, the RTP is handled by * itself. I guess this would work if both Alice and Bob were NAT'ed on the inside of the same NAT box. The problem is that if Alice and Bob both have NAT=yes and CANREINVITE=yes and they're on separate NAT'ed networks, the call is broken. So it's a dangerous configuration. If someone made a solution that * Compared the inside address AND the outside (NAT public IP) * If they are similar (NAT from the same network and public IP equals), connect the RDP streams from inside NAT to inside NAT However, with STUN, the calee or the caller might not present the inside IP address and therefore this will not be possible at all... Better to have an outbound SIP proxy that could make this happen. Or? /O ___ 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] P2P RTP without SIP re-invites
hi I guess this would work if both Alice and Bob were NAT'ed on the inside of the same NAT box. The problem is that if Alice and Bob both have NAT=yes and CANREINVITE=yes and they're on separate NAT'ed networks, the call is broken. So it's a dangerous configuration. nope. I have a public * server (beta server for a free VoIP service), on a public IP. and some sip phones around , like one in my home, behind nat, one in my office (another nat) and some others at my coworkers home... all behind nat. and are different nat box, do you agree? that works ok, I have RTP passing directly from one endpoint to the other... no RTP on the public * server. No stun is used. The phones are budgetones in this case. All are configured with nat=yes on asterisk side. or I missing something? -- Brancaleoni Matteo [EMAIL PROTECTED] Espia - Emmegi Srl ___ 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] availability of the ExtensionState was Asterisk Manager Interface notes
mattf wrote: Hello, I was referring to the availability of the ExtensionState Action (see the wiki: http://www.voip-info.org/tiki-index.php?page=Asterisk%20manager%20API ), even though I don't actually use it. Ok. Has anyone used theExtensionState Action sucessfully? I can't get it to work. For my purposes of status of an extension I wrote an updater script, that runs outside of Astrisk, every second and grabs the Action: Command Command: Show Channels output and parses it into a database so that my GUI applications can see which extensions and zap channels are busy and what they are connected to. It's not the most elegant solution, but it doesn't drain resources and I've been using it for months so it serves it's purpose. MATT--- Thanks for the tip. Julio Anjos Portugal ___ 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] determining legal VoIP service
I heard this argument when in SA by a group using Net2Phone- it was not accepted by the head of the Foreign Department of Telkom with whom I also spoke. Also of importance is that in many countries there have been police raids and confiscation of equipment used by CallShops because the CallShops are deemed by the incumbent to be 'breaking out'. You by offering these services would be encouraging them to do so. For consumers their task is impossible. At this point it is mostly about MONEY - if the equipment used is cheaply priced it may be worthwhile for your customers to just have it raided once in a while or if your rates so good they can afford the bribes to keep the police out. As far as your legal position it comes down to TWO things: 1. What that state says 2. Their ability to enforce it With Spain it is part of the European Union and Free Competition legilslation so there is no issue at all. With Nigeria it is illegal however even if their courts obtain judgement against you (probably without you knowing) they would be set upon the task of taking the judgement to your juristiction to enforce it - if where you live has a bilateral agreement with that country. Final comment - there are a lot of people doing what you want to out of Nigeria. I even know of one guy who is married to a barrister. I don't think you will have a problem finding people who will want to do the business. Stephen Wingfield Actually I believe this is one of the few things that can be done without worrying about the state(s) PUC coming down on your head. Since your users are in another country the state PUC cannot consider you providing a telephone service in their jurisdiction. - Dustin - Walker Haddock wrote: Can anyone recommend who we can consult with that could provide advice on the legality of a proposed VoIP service. Specifically, we would provide VoIP termination in the USA to clients in Spain, Nigeria and Guana. The termination service would connect the VoIP clients to the PSTN through a carrier like MCI, Verizon, etc. The calls placed would connect anywhere in the world via the USA carrier. Thanks, Walker ___ 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] newbie thinclient env
On Saturday 31 January 2004 12:13, you wrote: Hi! - asterix on the LTSP servers ? It is not advisable to run Asterisk on a server that runs X-Windows. You could give it a try on one machine, but I might very well turn out that you'll not like the resulting voice quality (choppy sound). - can i connect a phone to the thinclients via USB ( some thinclients have no PCI extension possibilities ) First thing is to check with LTSP if it supports local USB devices. And have you arranged already arranged sound support for the clients at all? Anyway, I'd rather go for a standard hardware IP phone - unless you are running a call center. Can you explain a bit more about that hardware IP phone. Maybe a link ? Could i just connect that to the network ? USB support is working already, the sound-chip, i must check on that. I think the newer LTSP-4 would be ideally for that, because you can configure some stuff to run locally. Should try that some day. - 17 LTSP servers have dynamic ip addresses ( here in belgium, that means that every 1 or 2 days, the ip address changes ). Can i use dyndns for asterix ? I wouldn't rely on that, dyndns can be slow and out-of-date. Instead register the dynamic Asterisk boxes with one or two static Asterisk boxes, which would basically mean that you'd end up with a star topolgy. - does asterix imply a serious load on the server, say for 5 telephones That depends on the codecs that you plan to use (which in turn depends on the phones that you use), as well as the number of concurrent calls. If transcoding is involved (=translation from one codec to another) then your CPU will be less bored. You should also consider QoS issues when running VoIP in a LTSP environment (for the record: Linux Terminx Server Project). I was thinkingh the use max 1 phone /thinclient, so for the moment beingh, max 5 phones / LTSP server. But yes, maybe it is too much for that one server, but i could easily put another one nearby it. Finally: Do you need your telephones to work even when the local LTSP server is down? Never thought about that, but i think on every location, there will still be one classical phone . Check the Wiki for codec details, bandwidth usage etc: http://www.voip-info.org/tiki-index.php?page=Asterisk I surelly do that. Thanks for the info ! Jef ___ 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] asterisk php status viewer
since I was annoyed this morning, I wrote this simple php script to output channel status from asterisk manager. disclaimer that's very bad written, nor commented... I wrote that just for fun /disclaimer and if someone will use that / improve it , just lemme know. http://asterisk.espia-net.net (wrote with php 4.3.3 and depends on Event: StatusComplete, so a recent * cvs version is needed) -- Brancaleoni Matteo [EMAIL PROTECTED] Espia - Emmegi Srl ___ 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] smtp question
The headers From:, Reply-To: etc generally ARE things like MAILER-DAEMON. The envelope sender used in the SMTP conversation and Return-Path: should be . On Sat, Jan 31, 2004 at 01:20:02AM -0600, Brian West said: Correct me if I'm wrong here but when a message bounces and the mailer/mta generates a bounce message shouldn't the from field have in it instead of an email addres (ie. [EMAIL PROTECTED]). The list was nailed with over 13,000 bounce messages(and they keep coming) from ONE list subscriber and I wasn't sure if their smtp server wasn't doing what it should. I finally had to just block their domain totally at postini to stop the flood of messages. I think this was part of the first slowdown and why the list slowed down again today. ___ 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] SIP gateway question
Just received a Mediatrix 1204 fxo sip gateway and playing with the initial config's, etc. It's working, but have a ways to go before it could be considered usable. The box was not designed to register like sip phones do. The incoming pstn line is an ordinary 2-wire analog US pots line, and I'm using canreinvite=no to forcably keep * in the middle for now. Questions: 1. The 1204 answers incoming pstn calls correctly, cycles through the invite/ trying/ringing (I have * config'ed to simply ring an internal sip phone for testing purposes), and I answer the call just fine from the sip phone. When I hang up the sip phone, * sends a Bye and the 1204 says OK. The 1204 then sends one more packet to * with both the source and destination ports one digit greater then what was used for the rtp session. I'm assuming that's a bug in their code; anyone seen something like that before? 2. The 1204 seems to be set to a 30 millisecond sampling rate while all other sip phones, etc, are set to 20. Anyone have any thoughts as to whether that would cause a problem later, or should I change that to 20 milliseconds for consistency? 3. Has anyone played with this box and found any unusual problems, weird config's, etc? The box is essentially in a test/eval mode, anticipating using it to replace a couple of x100p's. Rich ___ 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] determining legal VoIP service
On Fri, 30 Jan 2004, Dustin Goodwin wrote: Actually I believe this is one of the few things that can be done without worrying about the state(s) PUC coming down on your head. Since your users are in another country the state PUC cannot consider you providing a telephone service in their jurisdiction. On the other hand, this is quite likely not allowed on the Nigeria and Ghana ends. That's the way it is in my country - South Africa - anyway. Steve ___ 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] Internal Lines Dialing Out
On Fri, 30 Jan 2004, Steve Rodgers wrote: Oops! I forgot the leading underscore. Use this version below. Steve. exten =_ NXX,1,Dial(Zap/1/$EXTEN) exten = _1NXXNXX,1,Dial(Zap/1/$EXTEN) And reaching us wot is in the rest of the world...? ;-) Steve ___ 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] determining legal VoIP service
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Walker Haddock Sent: Friday, January 30, 2004 5:52 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] determining legal VoIP service Can anyone recommend who we can consult with that could provide advice on the legality of a proposed VoIP service. Specifically, we would provide VoIP termination in the USA to clients in Spain, Nigeria and Guana. The termination service would connect the VoIP clients to the PSTN through a carrier like MCI, Verizon, etc. The calls placed would connect anywhere in the world via the USA carrier. If you're interested in receiving traffic for those locations, you could talk to ITXC. They mostly sell H.323 termination though other people's POP sites around the world. They may need more termination in those areas. If so, they are very well versed in this type of thing, and could probably help you out (not just in getting your proosed plan running, but possibly making a good bit of money on top of that). I don't know anyone in the reseller/sales division, only their engineers, but you might want to give them a try. The could become a very good customer/peer of yours if you happen to be terminating in the right spots. http://www.itxc.net FYI, I don't see much on their web site about becoming a SNOC as they call it, but I'd try just giving sales or a general number a call and see who you can get transferred to. Best of luck, Daryl G. Jurbala BMPC Network Operations Tel (NY): +1 917 477 0468 x235 Tel (MI): +1 616 608 0004 x235 Tel (UK): +44 208 792 6813 x235 Fax: +1 508 526 8500 INOC-DBA: 26412*DGJ PGP Key: http://www.introspect.net/pgp ___ 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] Caller ID Presentment on PRI...
Hey folks, I have a T100P card with a PRI; when doing outbound dialing over the PRI, I can use SetCIDNum(2024561414|a) to force caller ID to display as The White House on a land line. This is apparently done as a reverse lookup by Verizon, as I do not hand the PRI the words The White House -- 202-456-1414 is an actual White House number. Using SetCIDName(Flying Pigs) makes no difference to what is displayed on the remote landline. It still says The White House or whatever the reverse lookup is for the number supplied. What I am wondering is whether a PRI has to have some feature turned on in order to inject the textual Caller ID name data, or whether this is just how PRI's work in general. And if so, who controls the reverse mapping, and how does that work with CLEC phone numbers, etc? Is anybody out there currently able to set CIDName to be something different than the reverse lookup name? My goal is not to spoof the White House, btw, but it makes a fun example. Thanks, Dave = David C. Troy [EMAIL PROTECTED] 410-384-2500 Sales ToadNet - Want to go fast?410-544-1329 FAX 570 Ritchie Highway, Severna Park, MD 21146-2925 www.toad.net ___ 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] Caller ID Presentment on PRI...
What I am wondering is whether a PRI has to have some feature turned on in order to inject the textual Caller ID name data, or whether this is just how PRI's work in general. And if so, who controls the reverse mapping, and how does that work with CLEC phone numbers, etc? Nope -- that's how PSTN-connected PRIs work -- you send the #, they map the name and display that. The only way you're going to be able to set the textual CID information is if you're trunking to your own equipment (i.e. trunking to a Nortel Meridian system or something to that effect) Regards, Andrew ___ 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] Caller ID Presentment on PRI...
On Sat, 31 Jan 2004, David C. Troy wrote: What I am wondering is whether a PRI has to have some feature turned on in order to inject the textual Caller ID name data, or whether this is just how PRI's work in general. And if so, who controls the reverse mapping, and how does that work with CLEC phone numbers, etc? You'll need to get in touch with whomever provides your PRI to change that. It's often a somewhat interesting process, and will only affect specific phone numbers -- the name doesn't travel with the call, it's retrieved as a seperate operation, and the request is routed back based off of the phone number. You can change your phone numbers to say whatever, but the White House's phone numbers can't be affected by you. Is anybody out there currently able to set CIDName to be something different than the reverse lookup name? My goal is not to spoof the White House, btw, but it makes a fun example. Beware the three-letter agencies. Beware even more the two-letter ones. -rt ___ 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] How to delay dialing
First, turn off your HTML in all posts to the list.
On Saturday 31 January 2004 02:03, Terence Parker wrote:
br /Thanks again for the patch - and the updated patch too!
br /
br /Actually, I was looking more closely at the chan_vpb.c file
earlier br /after the first patch and tried recompiling * again
after removing br /'char' from char t* - since I gather that wasn't
required again. br /Asterisk now compiles properly, but crashes
when a call is actually br /made out (with and without the 'w').
Here's what happens: br /
br /With the extensions setting of quot;exten =gt;
br /_9.,1,Dial(vpb/1-1/w${EXTEN:1},r)quot; - when I dial a number,
for example br /918501, asterisk crashes with the output:
br /
br /-- Executing Dial(quot;SIP/TerenceParker-465dquot;,
quot;vpb/1-1/w18501|rquot;) in new br /stack
br / Read_channel ## vpb/1-1: Setting record mode, bridge = 0
br / -- 1-1 requested, got: [vpb/1-1]
br /Ouch ... error while writing audio data: : Broken pipe
br /Ouch ... error while writing audio data: : Broken pipe
br /Ouch ... error while writing audio data: : Broken pipe
br /
br /[1]+ Segmentation fault asterisk -vvvg
br /
br /Also, when I try to revert back to my standard dial plan using
quot;exten br /=gt; 9,1,Dial(vpb/1-1/)quot; - asterisk crashes
in exactly the same way, br /except without the 'w'.
br /
br /Though I don't program in C, reading through your patch it
looks fine br /to me - I can see that you're merely trying to find
instances of 'w' or br /'f' after the / and replace them with , or
amp; . What I don't know is how br /the rest of the chan_vpb.c
file interacts with that function. br /
br /Any further ideas on how to solve this?
I'll look at it again, but since I don't have a VoiceTronix card
installed in any of my machines yet, I can't test it directly.
-Tilghman
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RE: [Asterisk-Users] Music on Hold Warnings
Tilghman Thanks for the help. You were spot on, yup the bitrate was screwed. NOTICE[1100258240]: res_musiconhold.c:260 monmp3thread: Request to schedule in the past?!?! And the machine does seem to be heavily underload - Asterisk = 100% CPU. MOH is working great now. Thanks. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tilghman Lesher Sent: 30 January 2004 16:39 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Music on Hold Warnings On Friday 30 January 2004 04:33, Craig Waddington wrote: 1.Warning, flexibel rate not heavily tested! You're using variable rate mp3's. If you want to avoid the error, recode your mp3s to a static rate. 2. NOTICE[1100258240]: res_musiconhold.c:260 monmp3thread: Request to schedule in the past?!?! Is your machine heavily loaded? This could indicate that a thread was unable to complete a task because it was interrupted and did not resume for a fairly long time (as processor time goes). It could also indicate clock drift (sync your time with NTP servers more often). -Tilghman ___ 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
[Asterisk-Users] TE410P E1 PRI problem
Hi everyone!
Here is my configuration and messages taken from
Asterisk startup. The E1 PRI trunk is connected to our national telecom company
here in Croatia. When I call from outside over this trunk to my company I get
'error in connection' respnse. In the same moment I can't see anything in
Asterisk, nothing that will tell me that the call reached Asterisk.I think
there is a problem with PRI synchronization or PRI to zap
communication.
The card I am using is TE410P, the first port is
the one that I use.
/etc/zaptel.conf
-
# port 1: trunk to telecom
span=1,0,0,ccs,hdb3bchan=1-15,17-31dchan=16
After loading module ztcfg shows:
--
[EMAIL PROTECTED] asterisk]# /sbin/ztcfg
-v
Zaptel
Configuration==
SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet
(DSX-1)
31 channels configured.
/etc/asterisk/zapata.conf
; trunk:switchtype = euroisdnsignalling =
pri_cpegroup = 2context = defaultchannel =
1-15,17-31
extensions.conf
---
I have demo context section included in default
context where incoming calls from PRI trunk are terminated. (When I dial from
SIP phone terminated in same default context I get the response., so I think
there is no problem with context)
During Asterisk startup I get this output
regarding Zap channel
---
[chan_zap.so] = (Zapata
Telephony w/PRI) == Parsing '/etc/asterisk/zapata.conf': FoundJan
31 17:04:41 WARNING[1074441696]: chan_zap.c:7552 setup_zap: Ignoring
rxwink -- Registered channel 1, PRI Signalling
signalling -- Registered channel 2, PRI Signalling
signalling -- Registered channel 3, PRI Signalling
signalling -- Registered channel 4, PRI Signalling
signalling -- Registered channel 5, PRI Signalling
signalling -- Registered channel 6, PRI Signalling
signalling -- Registered channel 7, PRI Signalling
signalling -- Registered channel 8, PRI Signalling
signalling -- Registered channel 9, PRI Signalling
signalling -- Registered channel 10, PRI Signalling
signalling -- Registered channel 11, PRI Signalling
signalling -- Registered channel 12, PRI Signalling
signalling -- Registered channel 13, PRI Signalling
signalling -- Registered channel 14, PRI Signalling
signalling -- Registered channel 15, PRI Signalling
signalling -- Registered channel 17, PRI Signalling
signalling -- Registered channel 18, PRI Signalling
signalling -- Registered channel 19, PRI Signalling
signalling -- Registered channel 20, PRI Signalling
signalling -- Registered channel 21, PRI Signalling
signalling -- Registered channel 22, PRI Signalling
signalling -- Registered channel 23, PRI Signalling
signalling -- Registered channel 24, PRI Signalling
signalling -- Registered channel 25, PRI Signalling
signalling -- Registered channel 26, PRI Signalling
signalling -- Registered channel 27, PRI Signalling
signalling -- Registered channel 28, PRI Signalling
signalling -- Registered channel 29, PRI Signalling
signalling -- Registered channel 30, PRI Signalling
signalling -- Registered channel 31, PRI Signalling
signalling == Starting D-Channel on span 1 == Registered
channel type 'Zap' (Zapata Telephony Driver w/PRI) == Registered
channel type 'Tor' (Zapata Telephony Driver w/PRI) == Registered
application 'CallingPres' == Manager registered action
ZapTransfer == Manager registered action ZapHangup ==
Manager registered action ZapDialOffhook
If I turn on pri debugging in Asterisk CLI (pri
intense debug span 1) I get this every second:
--
[00 01 7f ] Unnumbered
frame: SAPI: 00 C/R: 0 EA: 0 TEI:
000 EA: 1 M3:
3 P/F: 1 M2: 3 11: 3 [ SABME (set asynchronous balanced mode
extended) ] 0 bytes of dataSending Set Asynchronous Balanced Mode
Extended
The other direction, if I try to call out from
internal SIP phone:
-- Executing
Dial("SIP/7001-da6d", "Zap/g1/098227655") in new stackJan 31 17:36:20
NOTICE[1256866752]: app_dial.c:527 dial_exec: Unable to create channel
of type 'Zap' == Everyone is busy at this time
an anyone help, please!
Tomica Crnek
Re: [Asterisk-Users] How to delay dialing
I've got one I might be able to loan you for a couple of months, if you
have a great desire to hack away.
John
On Sat, 2004-01-31 at 10:21, Tilghman Lesher wrote:
First, turn off your HTML in all posts to the list.
On Saturday 31 January 2004 02:03, Terence Parker wrote:
br /Thanks again for the patch - and the updated patch too!
br /
br /Actually, I was looking more closely at the chan_vpb.c file
earlier br /after the first patch and tried recompiling * again
after removing br /'char' from char t* - since I gather that wasn't
required again. br /Asterisk now compiles properly, but crashes
when a call is actually br /made out (with and without the 'w').
Here's what happens: br /
br /With the extensions setting of quot;exten =gt;
br /_9.,1,Dial(vpb/1-1/w${EXTEN:1},r)quot; - when I dial a number,
for example br /918501, asterisk crashes with the output:
br /
br /-- Executing Dial(quot;SIP/TerenceParker-465dquot;,
quot;vpb/1-1/w18501|rquot;) in new br /stack
br / Read_channel ## vpb/1-1: Setting record mode, bridge = 0
br / -- 1-1 requested, got: [vpb/1-1]
br /Ouch ... error while writing audio data: : Broken pipe
br /Ouch ... error while writing audio data: : Broken pipe
br /Ouch ... error while writing audio data: : Broken pipe
br /
br /[1]+ Segmentation fault asterisk -vvvg
br /
br /Also, when I try to revert back to my standard dial plan using
quot;exten br /=gt; 9,1,Dial(vpb/1-1/)quot; - asterisk crashes
in exactly the same way, br /except without the 'w'.
br /
br /Though I don't program in C, reading through your patch it
looks fine br /to me - I can see that you're merely trying to find
instances of 'w' or br /'f' after the / and replace them with , or
amp; . What I don't know is how br /the rest of the chan_vpb.c
file interacts with that function. br /
br /Any further ideas on how to solve this?
I'll look at it again, but since I don't have a VoiceTronix card
installed in any of my machines yet, I can't test it directly.
-Tilghman
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Re: [Asterisk-Users] smtp question
Yep thats what I was thinking but apparently that isn't what is going on. We have have 41,000 bounces that have been blocked from said mail server. The from is the mail server and no return path is set. EVIL bkw On Sat, 31 Jan 2004, Walt Reed wrote: The headers From:, Reply-To: etc generally ARE things like MAILER-DAEMON. The envelope sender used in the SMTP conversation and Return-Path: should be . On Sat, Jan 31, 2004 at 01:20:02AM -0600, Brian West said: Correct me if I'm wrong here but when a message bounces and the mailer/mta generates a bounce message shouldn't the from field have in it instead of an email addres (ie. [EMAIL PROTECTED]). The list was nailed with over 13,000 bounce messages(and they keep coming) from ONE list subscriber and I wasn't sure if their smtp server wasn't doing what it should. I finally had to just block their domain totally at postini to stop the flood of messages. I think this was part of the first slowdown and why the list slowed down again today. ___ 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] asterisk php status viewer
Looks interesting I will check it out and see what I can do with it =) On Sat, 2004-01-31 at 08:17, Brancaleoni Matteo wrote: since I was annoyed this morning, I wrote this simple php script to output channel status from asterisk manager. disclaimer that's very bad written, nor commented... I wrote that just for fun /disclaimer and if someone will use that / improve it , just lemme know. http://asterisk.espia-net.net (wrote with php 4.3.3 and depends on Event: StatusComplete, so a recent * cvs version is needed) ___ 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] How to delay dialing
Oops... sorry about that, stupid webmail system defaults to HTML (don't use it very often so frequently overlook this. I usually send mail using 'apple mail', which seems to screw up even plain text e-mails, but... well... have to retaliate against Outlook Express users some way!! Anyways - does anyone know if Voicetronix even supports the use of a 'comma' or '' even after they are successfully converted from 'w' and 'f' respectively? Thanks again. Terence I'll look at it again, but since I don't have a VoiceTronix card installed in any of my machines yet, I can't test it directly. -Tilghman ___ 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
[Asterisk-Users] Are there any list moderators?
I'm just curious... I have several times posted a message accidentally using the wrong account - since the address I use for this list isn't my default one. I often re-post using the correct account, and get a notification on the first that my message is 'pending approval'. I don't expect my wrong messages to get approved, but they don't seem to get rejected either - nor have I been sworn at by any moderators complaining about me wasting their time! ... does anyone actually moderate these such messages? Terence ___ 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] Multiple Line Appearances
How were you able to integrate this with asterisk? Or did you drop asterisk in favor of ser? John On Thu, 2004-01-29 at 12:44, John Todd wrote: At 12:20 PM -0500 1/29/04, Jeremy McNamara wrote: MLS Drop for SysAdmin wrote: Has anyone successfully implemented concurrent appearance of the same PBX extension on multiple SIP phones? When using Cisco 7960s under call manager, you can have several phones with the same line appearance, but the first user to seize a line makes it inaccessible to other phones. Under SIP operation it seems as though this is not possible, but we don't see group ringing definable for SIP extensions. It is my understanding that Cisco didn't bother implementing this functionality into their SIP firmware. However, as you have described, this feature does work when using CCM. chan_skinny (and chan_sccp - which btw, will become the same channel driver soon) will eventually support this feature. If you (anyone?) have any motivation for Theo and myself to make Asterisk's SCCP support go to the top of our to-do lists, please contact either one of us off-list. Jeremy McNamara The Cisco phones with SIP support this just fine; it's not a problem for the endpoints, it's a problem for the SIP registrar. The phones will happily send out the same authentication name/password pair all day long to the server. The server must be smart enough to then map those multiple registrations to a single number. Asterisk does not support this feature at this time. If you want to use this trick, try SER, as I have had multiple devices with the same registration data register against SER. When an INVITE is passed into the system, all the phones automatically ring and the first pickup gets the call. Typically, you'd want to do this type of multi-number mapping back on the server, anyway - it's not a good idea to have multiple endpoints registering with the same auth data, but you can do it if you really, really want - just not with Asterisk. JT ___ 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] Re: Grandstream Firmware ?
Yes. We're currently testing 1.04.45 before making it available on our web site (www.telappliant.com/grandstream). Tan telappliant.com - Original Message - From: Mike Machado [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, January 31, 2004 12:08 AM Subject: Re: [Asterisk-Users] Re: Grandstream Firmware ? Do both the budgetone and the handytone use the same firmware? On Fri, 2004-01-30 at 06:26, Stephen R. Besch wrote: Greg Boehnlein wrote: On Thu, 29 Jan 2004, Michael Welter wrote: I have 1.0.4.45 (beta) on my tftp server. Try it at 66.250.23.58. Cheers, Michael Welter Is there a changelog available for the Beta release train? I'm looking to see if they have fixed Early Dial yet. When GS connected to my * server to examine the problem, they promised to keep me posted on the early dial problem. I haven't heard anything yet, so I am assuming that it has not been fixed. ___ 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 ___ 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] SNOM 200 question
Are you sure that the snom isn't negotiating the GSM codec? I think that this is negotiated by default unless you have disallow/allow statement. To determine whether this is the problem, put the following into the [general] section of you sip.conf: disallow=all allow=ulaw allow=alaw As for the choppy sound on VM messages, i don't think you can do much about this. It's more down to the design than anything. Try putting the call on mute when listening to messages. Hope that helps. Tan telappliant.com - Original Message - From: Lane Hoskins [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, January 30, 2004 9:17 PM Subject: [Asterisk-Users] SNOM 200 question Question for other snom 200 users: 1. We have horrible sound quality regardless of the codec we use in the phone or specify in *. Has anyone else run into this early on and found a software fix? 2. Speakerphone will not work for playing VM messages, it chops the message into unintelligible fragments of audio. Any ideas? 3. Initially we have horrible introduction of background noise into the handset earpiece which seems to quiet after there is audio on the other end. Ideas? 4. Sound quality to called parties outside our system is intermittently horrible: static filled and raspy where we have to ask people to repeat themselves many times. Could this be related to powerline noise or something like that? We have 8 lines coming into our building. Two are the main lines which we have ringing to the receptionist first and then to selected other extens. This part works great. We need to map the keys on the SNOM 200 such that when there is a call on line 1 the top key flashes/lights steady depending on call state and any extension can pick it up even if it doesn't ring there by pressing the button. This needs to hold true for the 1st two lines, and one of the remaining 6 lines at each extension as we have direct dials. All calls come to * via a T1 Digium card and an Adtran TSU 600. There are 8 separate POTS lines to our building for voice. So in example - call comes in on pstn line 1 , button one flashes at all phones, someone answeres it, button one solid on all phones, call comes in on line 2, button 2 flashes on all phones,can be answered from anywhere by simply hitting that button, gets answered and button changes to solid on all phones, call comes to me from line 8 (my direct dial line) and button 3 flashes on my phone only (my phone will also ring b/c it's set up that way in the dialplan) I can put other caller on hold and answer line 8 simply by pressing the button. Is this an easy thing to do that I'm simply not seeing? Thanks, Lane Hoskins, MCP Network Engineer Automated Horizons Inc. Direct - 540.767.7626 Main - 540.767.7600 ___ 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] Words for Allison(?)
On Fri, Jan 30, 2004 at 10:48:35PM -0500, John Todd wrote: You may consider putting together concrete lists of words so that I or others may keep them on short lists so that when we have Allison do various recordings we can find them in a single place. In fact, a bugnote would be the optimal place to put them and then mail me with the bug ID #. I'm still trying to put that list together, and identify which words/phrases have suitable versions already in CVS. In the mean time, I've seen references to bug #'s, here on the list and in the CVS logs. I've yet to stumble across the bug tracking system, though -- can you give me a nudge in the right direction? Thanx, Rob -- Rob Fugina, Systems Guy [EMAIL PROTECTED] -- http://www.geekthing.com My firewall filters MS Office attachments. Free Tibet*! *With the purchase of any country of equal or greater value. ___ 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] Words for Allison(?)
bugs.digium.com On Sat, 2004-01-31 at 12:24, Rob Fugina wrote: On Fri, Jan 30, 2004 at 10:48:35PM -0500, John Todd wrote: You may consider putting together concrete lists of words so that I or others may keep them on short lists so that when we have Allison do various recordings we can find them in a single place. In fact, a bugnote would be the optimal place to put them and then mail me with the bug ID #. I'm still trying to put that list together, and identify which words/phrases have suitable versions already in CVS. In the mean time, I've seen references to bug #'s, here on the list and in the CVS logs. I've yet to stumble across the bug tracking system, though -- can you give me a nudge in the right direction? Thanx, Rob -- Go to http://www.digium.com/index.php?menu=documentation and look at the Unofficial Links section. This section has links to a wide variety of 3rd party Asterisk related pages. My page is the Asterisk Resource Pages. BTEL Consulting 504-899-1387 or 850-484-4545 or 877-677-9643 ___ 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] asterisk with big number of extentions.
On Sat, 2004-01-31 at 10:36, WipeOut wrote: Fran Boon wrote: OK, so what success have people had with which clustering technologies? I'm more interested in resilience than performance. I would think that failover clustering would be far easier than a load sharing or processing cluster.. Great, so that works for me :) For lots of info on various clustering a HA systems take a look at http://www.linux-ha.org/ This looks like a great resource :) Has anyone successfully used this with Asterisk? Cheers, F ___ 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] Words for Allison(?)
Rob Fugina said: On Fri, Jan 30, 2004 at 10:48:35PM -0500, John Todd wrote: In the mean time, I've seen references to bug #'s, here on the list and in the CVS logs. I've yet to stumble across the bug tracking system, though -- can you give me a nudge in the right direction? Thanx, Rob http://bugs.digium.com/ Its the first entry in the google result when you search for asterisk bug tracking!!! You may also want to check out http://www.asterisk.org and the documentation wiki at: http://www.voip-info.org/wiki-Asterisk if you havn't stumbled across them yet. Robert ___ 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] Words for Allison(?)
On Sat, 2004-01-31 at 18:24, Rob Fugina wrote: In the mean time, I've seen references to bug #'s, here on the list and in the CVS logs. I've yet to stumble across the bug tracking system, though -- can you give me a nudge in the right direction? http://bugs.digium.com/ F ___ 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] SIP gateway question
Rich Adamson wrote: The 1204 then sends one more packet to * with both the source and destination ports one digit greater then what was used for the rtp session. I'm assuming that's a bug in their code; anyone seen something like that before? That would be RTCP (RTP + 1) 3. Has anyone played with this box and found any unusual problems, weird config's, etc? I have several of these boxes in use at a few different sites. Once installed, I have never gone back in and looked at any of them. They just work. I have it running in canreinvite mode and all sip phones running p2p. The poor * box has really no work to do. -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ 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] SIP gateway question
Hi Bob, The 1204 then sends one more packet to * with both the source and destination ports one digit greater then what was used for the rtp session. I'm assuming that's a bug in their code; anyone seen something like that before? That would be RTCP (RTP + 1) 3. Has anyone played with this box and found any unusual problems, weird config's, etc? I have several of these boxes in use at a few different sites. Once installed, I have never gone back in and looked at any of them. They just work. I have it running in canreinvite mode and all sip phones running p2p. The poor * box has really no work to do. I'm trying to figure out how best to bring pstn calls into * using this box, and not sure I'm there yet. Since the box doesn't register with *, I'm using the Redirect method which effectively causes the 1204 to dial x3094. What I'd like to do is simply drop that incoming call into the ivr menu directly. Any thoughts on how best to do that? Rich ___ 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] Compiling while * is running
While your problem is most likely bad RAM as other replies have suggested, there is another thing to keep in mind. Some implementations of dynamic module loading have problems if a loaded module is overwritten on the disk. What this means is that it is safest to stop Asterisk just before running make install, else the running instance may mysteriously segfault at that point. /w -- /~\ The ASCII Ribbon Campaign \ /No HTML/RTF in email X No Word docs in email / \ Respect for open standards ___ 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] 8 lines - best approach
How about a 16 port FXO/FXS card (your choice of FXO/FXS modules or any mix of them) for $999. Will that be a good option? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tilghman Lesher Sent: Saturday, January 24, 2004 5:35 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] 8 lines - best approach On Friday 23 January 2004 12:18, Paul Mahler wrote: On Friday, January 23, 2004 at 8:04 AM, Steven Critchfield wrote: On Fri, 2004-01-23 at 09:30, Darren Martz wrote: I have 8 lines coming into an existing PBX system and am looking for a cost effective way to replace the existing system with Asterisk. We need some of the features in Asterisk, including its ability to support remote offices (long distance savings). At first glance this appears to require a T100P card and a channel bank, but that seems rather expensive. My estimated price on that would be roughly $2600 for 8 lines given that system - perhaps my estimate is way off Is there another way that is more cost effective? That number sounds about right. It is likely that it will be less, but budgeting that much for hardware is a good start. Do you have to continue to use the existing handsets? You should look at replacing the existing phones with SIP phones. He did say cost-effective. Last I checked, 24 SIP phones (unless they are Grandstreams) will cost far more than a channel bank. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Internal Lines Dialing Out
Thanks for the tips, i tried it though and i still get the same thing. basically what happens is I pick up the phone, hear dialtone, dial the number, get a slight pause, here dial tone again (when i would expect it to be dialing), and then I dial the # again and it works, it seems that it is passing me through to the external line rather than dialing my digits. Here is my zapata.conf and zaptel.conf with a small snippet of debug: -- Starting simple switch on 'Zap/2-1' -- Executing Dial(Zap/2-1, Zap/1/$EXTEN) in new stack -- Called 1/$EXTEN -- Zap/1-1 answered Zap/2-1 -- Attempting native bridge of Zap/2-1 and Zap/1-1 -- Hungup 'Zap/1-1' == Spawn extension (internallines, 4310817, 1) exited non-zero on 'Zap/2-1' -- Executing Hangup(Zap/2-1, ) in new stack == Spawn extension (internallines, h, 1) exited non-zero on 'Zap/2-1' -- Hungup 'Zap/2-1' -- Starting simple switch on 'Zap/2-1' -- Hungup 'Zap/2-1' [EMAIL PROTECTED] etc]# more zaptel.conf fxsks=1 fxols=2 loadzone=us defaultzone=us [EMAIL PROTECTED] asterisk]# more zapata.conf [channels] language=en context=from-analog signalling=fxs_ks usecallerid=yes threewaycalling=yes echocancel=yes echocancelwhenbridged=yes ;immediate=yes channel = 1 signalling=fxo_ls context=internallines ;immediate=yes mailbox=21 channel = 2 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Steve Rodgers Sent: Saturday, January 31, 2004 12:00 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Internal Lines Dialing Out Oops! I forgot the leading underscore. Use this version below. Steve. [always-out-pots] ;as generic as possible to allow all calls out other than local extensions which loads first above exten =_ NXX,1,Dial(Zap/1/$EXTEN) exten = _NXX,2,Goto(102) exten = _NXX,102,Congestion exten = _NXX,103,Hangup exten = _1NXXNXX,1,Dial(Zap/1/$EXTEN) exten = _1NXXNXX,2,Goto(102) exten = _1NXXNXX,102,Congestion exten = _1NXXNXX,103,Hangup On Friday 30 January 2004 21:51, Steve Rodgers wrote: Try replacing these lines: [always-out-pots] ;as generic as possible to allow all calls out other than local extensions which loads first above exten = _.,1,Dial(Zap/1/$EXTEN) exten = _.,2,Goto(102) exten = _.,102,Congestion exten = _.,103,Hangup with these: [always-out-pots] ;as generic as possible to allow all calls out other than local extensions which loads first above exten = NXX,1,Dial(Zap/1/$EXTEN) exten = NXX,2,Goto(102) exten = NXX,102,Congestion exten = NXX,103,Hangup exten = 1NXXNXX,1,Dial(Zap/1/$EXTEN) exten = 1NXXNXX,2,Goto(102) exten = 1NXXNXX,102,Congestion exten = 1NXXNXX,103,Hangup I believe your problem is that you are not specific enough in your extension matching criteria. Also, I would recommend that you change your extension numbers to something like 110,112,113,114,115 ... Etc. These are not likely to conflict with normal telephone numbers, at least in North America anyway. Steve. On Friday 30 January 2004 20:21, Bruce Marler wrote: * Gurus, I have been trying, with mixed results, to setup an * server as a pbx in my home. Internal dialing works great, sip phone to sip phone and 1 fxs phone to sip phones, as well as inward dialing ringing all extensions then going to vmail. All great. But, when I try to dial out I run into issues, I have taken a look at the docs and the wiki and none of the tips have solved my problems. I have 1 fxs port and 1 fxo port (both digium cards) and as I said dial in works from the pstn. I want to dial my local extensions, but also be able to dialout my fxo port for anything not local, adding a 9 to be able to dial out is not an option (wife and kids would be all messed up:) FYI, also, if i set immediate=yes in my zapata.conf i can get straight dial tone and dial out but that does me little good since i am trying to get the value of dialing ext to ext in the house. All help is truly appreciated. Here is my extensions.conf file [general] static=yes writeprotect=yes [internallines] ;sip phones and fxs port use this as their context include = local-extensions include = always-out-pots exten = h,1,Hangup exten = i,1,Congestion exten = i,2,Hangup [always-out-pots] ;as generic as possible to allow all calls out other than local extensions which loads first above exten = _.,1,Dial(Zap/1/$EXTEN) exten = _.,2,Goto(102) exten = _.,102,Congestion exten = _.,103,Hangup [local-extensions] exten = 20,1,Dial(Zap/2-1,20) exten = 20,2,Voicemail(u21) exten = 20,102,Voicemail(b21) exten = 20,103,Hangup exten = 21,1,Dial(SIP/21,20) exten = 21,2,Voicemail(u21) exten = 21,102,Voicemail(b21) exten = 21,103,Hangup exten = 22,1,Dial(SIP/22,20) exten = 22,2,Voicemail(u21) exten = 22,102,Voicemail(b21) exten =
[Asterisk-Users] Dial app does not indicate ringing to calling party
I hope somebody has seen this before...
I'm trying to use a Dial command on a inbound call to ring multiple
destinations.The calls come in to me from the provider on IAX2, and one
of the destinations I try to ring is a IAX2 to call to my cell phone.
When I add the IAX2 destination into the Dial command, the setup I am trying
to achieve works (i.e. my Zap, SIP, and cell phone all ring) but the calling
party does not hear any ringing indication... just dead air until something
answers the call (voicemail or a human).
My IAX2 provider sends me all 10 digits so I have a context set up to catch
it and divert the call to my extension (7000) for handling:
exten = 2065475023,1,SetCIDname(DID 5475023)
exten = 2065475023,2,Goto(stations|7000|1)
Ext 7000 is:
MARK=Zap/2SIP/markIAX2/[EMAIL PROTECTED]/1206xxx
exten = 7000,1,Macro(stdexten,7000,${MARK})
stdext macro:
exten = s,1,Dial(${ARG2}|30|r)
exten = s,2,VoiceMail2([EMAIL PROTECTED])
exten = s,3,Hangup
exten = s,102,Voicemail2([EMAIL PROTECTED]) ;voicemail busy path
exten = s,103,Hangup
The problem seems to key on the fact that I'm dialing another IAX2
destination during handling of the inbound IAX2 call. If I dial ext 7000
from inside (SIP or my Zap device) I hear ringing. If I remove the IAX2
destination from $MARK so that it only tries to ring local devices, it
works.. the calling party hears ringing indication as well.
Is there something I'm missing about IAX2 to IAX2 dialing like this?
Thanks!
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RE: [Asterisk-Users] Internal Lines Dialing Out
exten = _.,1,Dial(Zap/1/$EXTEN)
exten = _.,1,Dial(Zap/1/${EXTEN})
Gotta put the name of the variable in brackets for it to work.
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[Asterisk-Users] The future of VoIP regulation (in the US)
Listers, Some of you may find this link of interest http://www.phoneplusmag.com/hotnews/41h3083829.html It was pointed out to me by one of our clients. Howard White ___ 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] Internal Lines Dialing Out
It's ${EXTEN} not $EXTEN
On Sat, 2004-01-31 at 14:32, Bruce Marler wrote:
Thanks for the tips, i tried it though and i still get the same thing.
basically what happens is I pick up the phone, hear dialtone, dial the
number, get a slight pause, here dial tone again (when i would expect it to
be dialing), and then I dial the # again and it works, it seems that it is
passing me through to the external line rather than dialing my digits.
Here is my zapata.conf and zaptel.conf with a small snippet of debug:
-- Starting simple switch on 'Zap/2-1'
-- Executing Dial(Zap/2-1, Zap/1/$EXTEN) in new stack
-- Called 1/$EXTEN
-- Zap/1-1 answered Zap/2-1
-- Attempting native bridge of Zap/2-1 and Zap/1-1
-- Hungup 'Zap/1-1'
== Spawn extension (internallines, 4310817, 1) exited non-zero on
'Zap/2-1'
-- Executing Hangup(Zap/2-1, ) in new stack
== Spawn extension (internallines, h, 1) exited non-zero on 'Zap/2-1'
-- Hungup 'Zap/2-1'
-- Starting simple switch on 'Zap/2-1'
-- Hungup 'Zap/2-1'
[EMAIL PROTECTED] etc]# more zaptel.conf
fxsks=1
fxols=2
loadzone=us
defaultzone=us
[EMAIL PROTECTED] asterisk]# more zapata.conf
[channels]
language=en
context=from-analog
signalling=fxs_ks
usecallerid=yes
threewaycalling=yes
echocancel=yes
echocancelwhenbridged=yes
;immediate=yes
channel = 1
signalling=fxo_ls
context=internallines
;immediate=yes
mailbox=21
channel = 2
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Steve Rodgers
Sent: Saturday, January 31, 2004 12:00 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Internal Lines Dialing Out
Oops! I forgot the leading underscore. Use this version below.
Steve.
[always-out-pots]
;as generic as possible to allow all calls out other than local extensions
which loads first above
exten =_ NXX,1,Dial(Zap/1/$EXTEN)
exten = _NXX,2,Goto(102)
exten = _NXX,102,Congestion
exten = _NXX,103,Hangup
exten = _1NXXNXX,1,Dial(Zap/1/$EXTEN)
exten = _1NXXNXX,2,Goto(102)
exten = _1NXXNXX,102,Congestion
exten = _1NXXNXX,103,Hangup
On Friday 30 January 2004 21:51, Steve Rodgers wrote:
Try replacing these lines:
[always-out-pots]
;as generic as possible to allow all calls out other than local
extensions which loads first above
exten = _.,1,Dial(Zap/1/$EXTEN)
exten = _.,2,Goto(102)
exten = _.,102,Congestion
exten = _.,103,Hangup
with these:
[always-out-pots]
;as generic as possible to allow all calls out other than local
extensions
which loads first above
exten = NXX,1,Dial(Zap/1/$EXTEN)
exten = NXX,2,Goto(102)
exten = NXX,102,Congestion
exten = NXX,103,Hangup
exten = 1NXXNXX,1,Dial(Zap/1/$EXTEN)
exten = 1NXXNXX,2,Goto(102)
exten = 1NXXNXX,102,Congestion
exten = 1NXXNXX,103,Hangup
I believe your problem is that you are not specific enough in your
extension matching criteria.
Also,
I would recommend that you change your extension numbers to something like
110,112,113,114,115 ... Etc.
These are not likely to conflict with normal telephone numbers, at least
in
North America anyway.
Steve.
On Friday 30 January 2004 20:21, Bruce Marler wrote:
* Gurus,
I have been trying, with mixed results, to setup an * server as a pbx in
my home. Internal dialing works great, sip phone to sip phone and 1 fxs
phone to sip phones, as well as inward dialing ringing all extensions
then going to vmail. All great.
But, when I try to dial out I run into issues, I have taken a look at
the
docs and the wiki and none of the tips have solved my problems.
I have 1 fxs port and 1 fxo port (both digium cards) and as I said dial
in works from the pstn.
I want to dial my local extensions, but also be able to dialout my fxo
port for anything not local, adding a 9 to be able to dial out is not an
option (wife and kids would be all messed up:)
FYI, also, if i set immediate=yes in my zapata.conf i can get straight
dial tone and dial out but that does me little good since i am trying to
get the value of dialing ext to ext in the house.
All help is truly appreciated.
Here is my extensions.conf file
[general]
static=yes
writeprotect=yes
[internallines]
;sip phones and fxs port use this as their context
include = local-extensions
include = always-out-pots
exten = h,1,Hangup
exten = i,1,Congestion
exten = i,2,Hangup
[always-out-pots]
;as generic as possible to allow all calls out other than local
extensions which loads first above
exten = _.,1,Dial(Zap/1/$EXTEN)
exten = _.,2,Goto(102)
exten = _.,102,Congestion
exten = _.,103,Hangup
[local-extensions]
exten = 20,1,Dial(Zap/2-1,20)
exten = 20,2,Voicemail(u21)
exten = 20,102,Voicemail(b21)
Re: [Asterisk-Users] Internal Lines Dialing Out
As the prevous poster pointed out, replace all instances of $EXTEN with
${EXTEN} and it should start working for you.
Steve.
On Saturday 31 January 2004 12:32, Bruce Marler wrote:
Thanks for the tips, i tried it though and i still get the same thing.
basically what happens is I pick up the phone, hear dialtone, dial the
number, get a slight pause, here dial tone again (when i would expect it to
be dialing), and then I dial the # again and it works, it seems that it is
passing me through to the external line rather than dialing my digits.
Here is my zapata.conf and zaptel.conf with a small snippet of debug:
-- Starting simple switch on 'Zap/2-1'
-- Executing Dial(Zap/2-1, Zap/1/$EXTEN) in new stack
-- Called 1/$EXTEN
-- Zap/1-1 answered Zap/2-1
-- Attempting native bridge of Zap/2-1 and Zap/1-1
-- Hungup 'Zap/1-1'
== Spawn extension (internallines, 4310817, 1) exited non-zero on
'Zap/2-1'
-- Executing Hangup(Zap/2-1, ) in new stack
== Spawn extension (internallines, h, 1) exited non-zero on 'Zap/2-1'
-- Hungup 'Zap/2-1'
-- Starting simple switch on 'Zap/2-1'
-- Hungup 'Zap/2-1'
[EMAIL PROTECTED] etc]# more zaptel.conf
fxsks=1
fxols=2
loadzone=us
defaultzone=us
[EMAIL PROTECTED] asterisk]# more zapata.conf
[channels]
language=en
context=from-analog
signalling=fxs_ks
usecallerid=yes
threewaycalling=yes
echocancel=yes
echocancelwhenbridged=yes
;immediate=yes
channel = 1
signalling=fxo_ls
context=internallines
;immediate=yes
mailbox=21
channel = 2
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Steve Rodgers
Sent: Saturday, January 31, 2004 12:00 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Internal Lines Dialing Out
Oops! I forgot the leading underscore. Use this version below.
Steve.
[always-out-pots]
;as generic as possible to allow all calls out other than local extensions
which loads first above
exten =_ NXX,1,Dial(Zap/1/$EXTEN)
exten = _NXX,2,Goto(102)
exten = _NXX,102,Congestion
exten = _NXX,103,Hangup
exten = _1NXXNXX,1,Dial(Zap/1/$EXTEN)
exten = _1NXXNXX,2,Goto(102)
exten = _1NXXNXX,102,Congestion
exten = _1NXXNXX,103,Hangup
On Friday 30 January 2004 21:51, Steve Rodgers wrote:
Try replacing these lines:
[always-out-pots]
;as generic as possible to allow all calls out other than local
extensions which loads first above
exten = _.,1,Dial(Zap/1/$EXTEN)
exten = _.,2,Goto(102)
exten = _.,102,Congestion
exten = _.,103,Hangup
with these:
[always-out-pots]
;as generic as possible to allow all calls out other than local
extensions
which loads first above
exten = NXX,1,Dial(Zap/1/$EXTEN)
exten = NXX,2,Goto(102)
exten = NXX,102,Congestion
exten = NXX,103,Hangup
exten = 1NXXNXX,1,Dial(Zap/1/$EXTEN)
exten = 1NXXNXX,2,Goto(102)
exten = 1NXXNXX,102,Congestion
exten = 1NXXNXX,103,Hangup
I believe your problem is that you are not specific enough in your
extension matching criteria.
Also,
I would recommend that you change your extension numbers to something
like 110,112,113,114,115 ... Etc.
These are not likely to conflict with normal telephone numbers, at least
in
North America anyway.
Steve.
On Friday 30 January 2004 20:21, Bruce Marler wrote:
* Gurus,
I have been trying, with mixed results, to setup an * server as a pbx
in my home. Internal dialing works great, sip phone to sip phone and 1
fxs phone to sip phones, as well as inward dialing ringing all
extensions then going to vmail. All great.
But, when I try to dial out I run into issues, I have taken a look at
the
docs and the wiki and none of the tips have solved my problems.
I have 1 fxs port and 1 fxo port (both digium cards) and as I said dial
in works from the pstn.
I want to dial my local extensions, but also be able to dialout my fxo
port for anything not local, adding a 9 to be able to dial out is not
an option (wife and kids would be all messed up:)
FYI, also, if i set immediate=yes in my zapata.conf i can get straight
dial tone and dial out but that does me little good since i am trying
to get the value of dialing ext to ext in the house.
All help is truly appreciated.
Here is my extensions.conf file
[general]
static=yes
writeprotect=yes
[internallines]
;sip phones and fxs port use this as their context
include = local-extensions
include = always-out-pots
exten = h,1,Hangup
exten = i,1,Congestion
exten = i,2,Hangup
[always-out-pots]
;as generic as possible to allow all calls out other than local
extensions which loads first above
exten = _.,1,Dial(Zap/1/$EXTEN)
exten = _.,2,Goto(102)
exten = _.,102,Congestion
exten = _.,103,Hangup
[local-extensions]
[Asterisk-Users] echo cancellation disabled
Hello I get these entries in my event log Jan 31 19:21:08 gateway kernel: zaptel Disabled echo canceller because of tone (rx) on channel 1 Do I have to change anything for enable echo cancellation? Regards Deepak
Re: [Asterisk-Users] Caller ID Presentment on PRI...
Hiya,
Is anybody out there currently able to set CIDName to be something
different than the reverse lookup name? My goal is not to spoof the
White House, btw, but it makes a fun example.
Beware the three-letter agencies. Beware even more the two-letter ones.
yeah, impersonating the White House sounds like asking for trouble ;-]
On topic: on most networks in europe you can't even fake the number,
everything that does not belong to your BRI/PRI is simply rewritten to
the primary number. BUT there is a service called UUS1 (UserUserSignaling),
just some text which can sent with the setup-message. A few phones (like
Ascom or Tiptel)
can set and display this message.
@kapejod, is there a change for a chan_capi that can read/set the uus1
to/from
the ${CALLERIDNAME}...? A lot of (really big) pbxes (Meridian etc) also set
the name
of the caller...
Regards,
andreas
_
Gaming galore at http://xtramsn.co.nz/gaming !
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Re: [Asterisk-Users] 8 lines - best approach
What card would that be? I would be interested to test it out. David - Original Message - From: Asterisk [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, January 31, 2004 12:00 PM Subject: RE: [Asterisk-Users] 8 lines - best approach How about a 16 port FXO/FXS card (your choice of FXO/FXS modules or any mix of them) for $999. Will that be a good option? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tilghman Lesher Sent: Saturday, January 24, 2004 5:35 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] 8 lines - best approach On Friday 23 January 2004 12:18, Paul Mahler wrote: On Friday, January 23, 2004 at 8:04 AM, Steven Critchfield wrote: On Fri, 2004-01-23 at 09:30, Darren Martz wrote: I have 8 lines coming into an existing PBX system and am looking for a cost effective way to replace the existing system with Asterisk. We need some of the features in Asterisk, including its ability to support remote offices (long distance savings). At first glance this appears to require a T100P card and a channel bank, but that seems rather expensive. My estimated price on that would be roughly $2600 for 8 lines given that system - perhaps my estimate is way off Is there another way that is more cost effective? That number sounds about right. It is likely that it will be less, but budgeting that much for hardware is a good start. Do you have to continue to use the existing handsets? You should look at replacing the existing phones with SIP phones. He did say cost-effective. Last I checked, 24 SIP phones (unless they are Grandstreams) will cost far more than a channel bank. -Tilghman ___ 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 ___ 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] Internal Lines Dialing Out
Thanks to both who replied, it works!!!
I cannot believe i missed that, talk about being knocked down a couple
notches:)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Steve Rodgers
Sent: Saturday, January 31, 2004 3:19 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Internal Lines Dialing Out
As the prevous poster pointed out, replace all instances of $EXTEN with
${EXTEN} and it should start working for you.
Steve.
On Saturday 31 January 2004 12:32, Bruce Marler wrote:
Thanks for the tips, i tried it though and i still get the same thing.
basically what happens is I pick up the phone, hear dialtone, dial the
number, get a slight pause, here dial tone again (when i would expect it
to
be dialing), and then I dial the # again and it works, it seems that it is
passing me through to the external line rather than dialing my digits.
Here is my zapata.conf and zaptel.conf with a small snippet of debug:
-- Starting simple switch on 'Zap/2-1'
-- Executing Dial(Zap/2-1, Zap/1/$EXTEN) in new stack
-- Called 1/$EXTEN
-- Zap/1-1 answered Zap/2-1
-- Attempting native bridge of Zap/2-1 and Zap/1-1
-- Hungup 'Zap/1-1'
== Spawn extension (internallines, 4310817, 1) exited non-zero on
'Zap/2-1'
-- Executing Hangup(Zap/2-1, ) in new stack
== Spawn extension (internallines, h, 1) exited non-zero on 'Zap/2-1'
-- Hungup 'Zap/2-1'
-- Starting simple switch on 'Zap/2-1'
-- Hungup 'Zap/2-1'
[EMAIL PROTECTED] etc]# more zaptel.conf
fxsks=1
fxols=2
loadzone=us
defaultzone=us
[EMAIL PROTECTED] asterisk]# more zapata.conf
[channels]
language=en
context=from-analog
signalling=fxs_ks
usecallerid=yes
threewaycalling=yes
echocancel=yes
echocancelwhenbridged=yes
;immediate=yes
channel = 1
signalling=fxo_ls
context=internallines
;immediate=yes
mailbox=21
channel = 2
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Steve Rodgers
Sent: Saturday, January 31, 2004 12:00 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Internal Lines Dialing Out
Oops! I forgot the leading underscore. Use this version below.
Steve.
[always-out-pots]
;as generic as possible to allow all calls out other than local
extensions
which loads first above
exten =_ NXX,1,Dial(Zap/1/$EXTEN)
exten = _NXX,2,Goto(102)
exten = _NXX,102,Congestion
exten = _NXX,103,Hangup
exten = _1NXXNXX,1,Dial(Zap/1/$EXTEN)
exten = _1NXXNXX,2,Goto(102)
exten = _1NXXNXX,102,Congestion
exten = _1NXXNXX,103,Hangup
On Friday 30 January 2004 21:51, Steve Rodgers wrote:
Try replacing these lines:
[always-out-pots]
;as generic as possible to allow all calls out other than local
extensions which loads first above
exten = _.,1,Dial(Zap/1/$EXTEN)
exten = _.,2,Goto(102)
exten = _.,102,Congestion
exten = _.,103,Hangup
with these:
[always-out-pots]
;as generic as possible to allow all calls out other than local
extensions
which loads first above
exten = NXX,1,Dial(Zap/1/$EXTEN)
exten = NXX,2,Goto(102)
exten = NXX,102,Congestion
exten = NXX,103,Hangup
exten = 1NXXNXX,1,Dial(Zap/1/$EXTEN)
exten = 1NXXNXX,2,Goto(102)
exten = 1NXXNXX,102,Congestion
exten = 1NXXNXX,103,Hangup
I believe your problem is that you are not specific enough in your
extension matching criteria.
Also,
I would recommend that you change your extension numbers to something
like 110,112,113,114,115 ... Etc.
These are not likely to conflict with normal telephone numbers, at least
in
North America anyway.
Steve.
On Friday 30 January 2004 20:21, Bruce Marler wrote:
* Gurus,
I have been trying, with mixed results, to setup an * server as a pbx
in my home. Internal dialing works great, sip phone to sip phone and 1
fxs phone to sip phones, as well as inward dialing ringing all
extensions then going to vmail. All great.
But, when I try to dial out I run into issues, I have taken a look at
the
docs and the wiki and none of the tips have solved my problems.
I have 1 fxs port and 1 fxo port (both digium cards) and as I said
dial
in works from the pstn.
I want to dial my local extensions, but also be able to dialout my fxo
port for anything not local, adding a 9 to be able to dial out is not
an option (wife and kids would be all messed up:)
FYI, also, if i set immediate=yes in my zapata.conf i can get straight
dial tone and dial out but that does me little good since i am trying
to get the value of dialing ext to ext in the house.
All help is truly appreciated.
Here is my extensions.conf file
[general]
static=yes
writeprotect=yes
[internallines]
;sip phones and fxs port use this as their context
include = local-extensions
include = always-out-pots
[Asterisk-Users] rtp sound quality?
pstn - sip gw - * - C7960 When I dial into * via the pstn, I hear the ivr menu just fine (good quality). I press 3000 (valid extn), and I begin to hear ringing however the ring back is very very choppy. I answer the C7960, and speech is clear in both directions. Place the C7960 extn on hold, and the MOH is very choppy. Checking 'sip show channels' indicates both the sip gw and C7960 are using ulaw; Both are 100 meg on local LAN. Top suggests all processes running less then 1 or 2 percent. The choppy sound only happens on the sip gw (Mediatrix 1204). MOH between two C7960's works fine. MOH via x100p works fine. Tried canreinvite=no and yes; no difference. Using a packet sniffer, I see a ~200 millisecond delay about every 1/2 second or so (varys), but nothing within the trace to hint at a layer-2 problem. Anyone have any thoughts as to why ringback and MOH are choppy but conversations are fine? Anything else I can look at to isolate the issue? Rich ___ 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] Multiple Line Appearances
My point in my last paragraph was that you don't want to do this on the user-agent side; you want to control this on the server side. To use asterisk parlance: exten = 1234,1,Dial(SIP/janeSIP/bill) This means that when extension 1234 is called, that the (single) phones named jane and bill will ring. Those SIP peers are defined in sip.conf. While the SIP specification allows one to have multiple endpoints registering with the same authentication name and password (and thus, the same identity) this is IMHO not a good design, since there is no accountability for where calls actually went or came from. I suppose if you are a completely open network then this is OK, but anywhere that there are monetary expenses associated with calls, this will quickly lead to heartache and woe. In any case, Asterisk only recognizes the most recent registration that it has seen for a particular identity, so every few seconds you'd get a tug-of-war going on with multiple identities registering to the same sip peer entry. SER can do both methods (multiple phones mapped to 1 identity, or multiple phones mapped to multiple identities) but this is the Asterisk mailing list, not the SER mailing list. :-) JT At 12:00 PM -0600 1/31/04, John Baker wrote: How were you able to integrate this with asterisk? Or did you drop asterisk in favor of ser? John On Thu, 2004-01-29 at 12:44, John Todd wrote: At 12:20 PM -0500 1/29/04, Jeremy McNamara wrote: MLS Drop for SysAdmin wrote: Has anyone successfully implemented concurrent appearance of the same PBX extension on multiple SIP phones? When using Cisco 7960s under call manager, you can have several phones with the same line appearance, but the first user to seize a line makes it inaccessible to other phones. Under SIP operation it seems as though this is not possible, but we don't see group ringing definable for SIP extensions. It is my understanding that Cisco didn't bother implementing this functionality into their SIP firmware. However, as you have described, this feature does work when using CCM. chan_skinny (and chan_sccp - which btw, will become the same channel driver soon) will eventually support this feature. If you (anyone?) have any motivation for Theo and myself to make Asterisk's SCCP support go to the top of our to-do lists, please contact either one of us off-list. Jeremy McNamara The Cisco phones with SIP support this just fine; it's not a problem for the endpoints, it's a problem for the SIP registrar. The phones will happily send out the same authentication name/password pair all day long to the server. The server must be smart enough to then map those multiple registrations to a single number. Asterisk does not support this feature at this time. If you want to use this trick, try SER, as I have had multiple devices with the same registration data register against SER. When an INVITE is passed into the system, all the phones automatically ring and the first pickup gets the call. Typically, you'd want to do this type of multi-number mapping back on the server, anyway - it's not a good idea to have multiple endpoints registering with the same auth data, but you can do it if you really, really want - just not with Asterisk. JT ___ 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] rtp sound quality?
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson pstn - sip gw - * - C7960 When I dial into * via the pstn, I hear the ivr menu just fine (good quality). I press 3000 (valid extn), and I begin to hear ringing however the ring back is very very choppy. Where are you getting timing from? Zaptel device? Ztdummy? -Josh ___ 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] rtp sound quality?
pstn - sip gw - * - C7960 When I dial into * via the pstn, I hear the ivr menu just fine (good quality). I press 3000 (valid extn), and I begin to hear ringing however the ring back is very very choppy. Where are you getting timing from? Zaptel device? Ztdummy? The * system has a pair of x100p's installed (and working), so I'm assuming zaptel (unless timing requires the call to come through the x100p first). Using the MOH (as an example), music is very choppy; however, I've noticed that if I try to talk over the top of MOH, then MOH sounds fine. Is this really the old timing thingie here too? Rich ___ 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] TE410P E1 PRI problem
On Sat, 31 Jan 2004, Tomica Crnek waxed: Hi everyone! Here is my configuration and messages taken from Asterisk startup. The E1 PRI trunk is connected to our national telecom company here in Croatia. When I call from outside over this trunk to my company I get 'error in connection' respnse. In the same moment I can't see anything in Asterisk, nothing that will tell me that the call reached Asterisk. I think there is a problem with PRI synchronization or PRI to zap communication. The card I am using is TE410P, the first port is the one that I use. /etc/zaptel.conf - # port 1: trunk to telecom span=1,0,0,ccs,hdb3 Try this instead: span=1,1,0,ccs,hdb3 (Note the second 1, that sets your timing to the telecom.) --Chris -- Chris Maj cmaj_hat_freedomcorpse_hot_info Pronunciation Guide: Maj == May Fingerprint: 43D6 799C F6CF F920 6623 DC85 C8A3 CFFE F0DE C146 ___ 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] Using an additional modem to get CallerID information
There are two stages in the process of getting callerID information from a
standard modem, to be used in Asterisk. The first stage is actually
capturing the information from the modem, the second stage is importing the
captured data into Asterisk.
Capturing the caller ID details from the modem
I will presume at this stage, that you have a modem that supports caller ID
and it is installed and configured to work with your Linux box.
Here is my first script that reads the details in...
#!/usr/bin/perl
$PortName = /dev/ttyn00;
$PortObj = open(MODEM,$PortName) || die Can't open $PortName: $!\n;
while (1==1) {
local $/ = \n;
while ($line=MODEM) {
chomp;
if ($line =~ s/NMBR = //) {
open(OUTFILE, /usr/src/myperl/callerid.txt) or die Can't open
callerid.txt: $!;
print OUTFILE $line;
close OUTFILE;
};
}
}
depending on your setup, you'll need to amend the $portName variable to
point to the port that you've installed the modem on. You also may want to
change the path that the callerid.txt file is written to.
Once the script is written, used the chmod A+X callerid.pl to change the
mode so that the program can be executed.
Finally run the program with parameter, to spawn the program as a new
process.
Using the callerid.txt file in Asterisk
Once the callerid.pl file has captured the callerid data, the number needs
to be loaded into asterisk. This is done using AGI functions within
asterisk.
Firstly create a perl script as follows.
#!/usr/bin/perl
open(INFILE, /usr/src/myperl/callerid.txt) or die cannot open file;
if ($callerID=INFILE) {
print SET CALLERID $callerID};
close INFILE;
once created, this script should be placed in the AGI directory.
Finally add a line to your extensions.conf file to call this script, an
example line would be.
Exten=_.,1,agi,getcallerid.pl
Hopefully, this should now leave you with CID working !!
attachment: winmail.dat
[Asterisk-Users] SUBSCRIBE in chan_sip - anyone?
So, what hardware or use is the SUBSCRIBE method used for in chan_sip.c? I asked this question a while ago, and got resounding silence. Maybe someone who is better at de-tangling C code than I am could take a peek. JT ___ 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] SUBSCRIBE in chan_sip - anyone?
So, what hardware or use is the SUBSCRIBE method used for in chan_sip.c? I asked this question a while ago, and got resounding silence. Maybe someone who is better at de-tangling C code than I am could take a peek. John, Not sure, but seems to me it came in about the time Olle and Snom were talking, and Olle was working on sip-2 or something like that. Rich ___ 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] rtp sound quality?
pstn - sip gw - * - C7960 When I dial into * via the pstn, I hear the ivr menu just fine (good quality). I press 3000 (valid extn), and I begin to hear ringing however the ring back is very very choppy. I answer the C7960, and speech is clear in both directions. Place the C7960 extn on hold, and the MOH is very choppy. Checking 'sip show channels' indicates both the sip gw and C7960 are using ulaw; Both are 100 meg on local LAN. Top suggests all processes running less then 1 or 2 percent. The choppy sound only happens on the sip gw (Mediatrix 1204). MOH between two C7960's works fine. MOH via x100p works fine. Tried canreinvite=no and yes; no difference. Using a packet sniffer, I see a ~200 millisecond delay about every 1/2 second or so (varys), but nothing within the trace to hint at a layer-2 problem. Anyone have any thoughts as to why ringback and MOH are choppy but conversations are fine? Anything else I can look at to isolate the issue? You need to disable VAD on the 1204. The 1204 stops xmiting RTP to * if it does not detect any acoustic energy. * can not clock itself sending RTP packets. It relyes on receiving RTP packets for it's timing. Try singing along with your MOH and the choppiness should go away, or disable VAD, or fix * RTP driver. Thanks Bob, that fixed it. Any other hints/issues/default values that I should muck with, or is that about it? Seems like it works pretty good; excellent echo cancellation, etc. I haven't done anything with the box as yet for dialing outbound. Anything to be concerned with, special parameters, etc? Rich ___ 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] 8 lines - best approach
On Sat, Jan 31, 2004 at 12:00:49PM -0800, Asterisk wrote: How about a 16 port FXO/FXS card (your choice of FXO/FXS modules or any mix of them) for $999. Will that be a good option? I know (me) someone (me) who'd make a (me) really good (me) beta tester (me)... -- Rob Fugina, Systems Guy [EMAIL PROTECTED] -- http://www.geekthing.com My firewall filters MS Office attachments. Behind every good computer - is a jumble of wire. ___ 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] 8 lines - best approach
well quit with the suspense already and tell us who! :-) -Original Message- From: Rob Fugina [mailto:[EMAIL PROTECTED] Sent: Saturday, January 31, 2004 7:56 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] 8 lines - best approach On Sat, Jan 31, 2004 at 12:00:49PM -0800, Asterisk wrote: How about a 16 port FXO/FXS card (your choice of FXO/FXS modules or any mix of them) for $999. Will that be a good option? I know (me) someone (me) who'd make a (me) really good (me) beta tester (me)... -- Rob Fugina, Systems Guy [EMAIL PROTECTED] -- http://www.geekthing.com My firewall filters MS Office attachments. Behind every good computer - is a jumble of wire. ___ 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] 8 lines - best approach
Rob Fugina writes: On Sat, Jan 31, 2004 at 12:00:49PM -0800, Asterisk wrote: How about a 16 port FXO/FXS card (your choice of FXO/FXS modules or any mix of them) for $999. Will that be a good option? I know (me) someone (me) who'd make a (me) really good (me) beta tester (me)... I know someone else (me) who would buy them (me). -- Speed, it seems to me, provides the one genuinely modern pleasure. -- Aldous Huxley (1894 - 1963) ___ 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] Compiling while * is running
Nope I do make install all the time with asterisk running without ONE problem. bkw On Sat, 31 Jan 2004, William Waites wrote: While your problem is most likely bad RAM as other replies have suggested, there is another thing to keep in mind. Some implementations of dynamic module loading have problems if a loaded module is overwritten on the disk. What this means is that it is safest to stop Asterisk just before running make install, else the running instance may mysteriously segfault at that point. /w -- /~\ The ASCII Ribbon Campaign \ /No HTML/RTF in email X No Word docs in email / \ Respect for open standards ___ 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] rtp sound quality?
Hi Rich! Anyone have any thoughts as to why ringback and MOH are choppy but conversations are fine? Anything else I can look at to isolate the issue? First guess (rather likely): Silence supression Second guess (unlikely): Non optimal Voice frames per TX as it is called in Grandstream setup; don't have a Mediatrix, so I can only guess. Should be 2 for the GS. Cheers, Philipp ___ 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] 8 lines - best approach
Thanks for the overwhelming response guys. Just wait for some time (6 weeks) and some of you will get to test it for sure. Watch out this mailing list for the announcement. For now, let's keep it a little secret. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jim Thompson Sent: Saturday, January 31, 2004 5:17 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] 8 lines - best approach Rob Fugina writes: On Sat, Jan 31, 2004 at 12:00:49PM -0800, Asterisk wrote: How about a 16 port FXO/FXS card (your choice of FXO/FXS modules or any mix of them) for $999. Will that be a good option? I know (me) someone (me) who'd make a (me) really good (me) beta tester (me)... I know someone else (me) who would buy them (me). -- Speed, it seems to me, provides the one genuinely modern pleasure. -- Aldous Huxley (1894 - 1963) ___ 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] Re: retrans_pkt: Maximum retries exceeded on call
I noticed this too and it is a pain to look at. I saw it because some of my SIP phones were turned off and the NOTIFY's for no voicemail reached maximum re-transmissions. Duh! Nobody was there to answer it. I didn't check to see what the log level was but if it only shows up on -vvv console option, I can live with it. Christian Hecimovic wrote: Try setting canreinvite=no in sip.conf. It might be that attempts to natively bridge the voice streams are failing. On Saturday 24 January 2004 23:26, Chris Wilson wrote: 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 ___ 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] Re: retrans_pkt: Maximum retries exceeded on call
That sending notify to endpoints that aren't registered has been fixed recently but we have one more bug that causes 0.0.0.0 bkw On Sat, 31 Jan 2004, Clif Jones wrote: I noticed this too and it is a pain to look at. I saw it because some of my SIP phones were turned off and the NOTIFY's for no voicemail reached maximum re-transmissions. Duh! Nobody was there to answer it. I didn't check to see what the log level was but if it only shows up on -vvv console option, I can live with it. Christian Hecimovic wrote: Try setting canreinvite=no in sip.conf. It might be that attempts to natively bridge the voice streams are failing. On Saturday 24 January 2004 23:26, Chris Wilson wrote: 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 ___ 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 ___ 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] Re: retrans_pkt: Maximum retries exceeded on call
2004-01-26 14:12 markster * channels/chan_sip.c (1.284): Don't send VMWI when we're not registered Yes that was fixed on the 26th. On Sat, 31 Jan 2004, Clif Jones wrote: I noticed this too and it is a pain to look at. I saw it because some of my SIP phones were turned off and the NOTIFY's for no voicemail reached maximum re-transmissions. Duh! Nobody was there to answer it. I didn't check to see what the log level was but if it only shows up on -vvv console option, I can live with it. Christian Hecimovic wrote: Try setting canreinvite=no in sip.conf. It might be that attempts to natively bridge the voice streams are failing. On Saturday 24 January 2004 23:26, Chris Wilson wrote: 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 ___ 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 ___ 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] Dial via sip gateway?
I'm having a brain fart
What's the proper syntax for dialing out via a sip g/w (Mediatrix)?
Been trying stuff similar to:
exten = _6X.,1,Dial(SIP/[EMAIL PROTECTED]/${EXTEN-1})
where 3091 is alias for the port on the Mediatrix. Sniffer indicates * did
even try the IP.
Rich
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[Asterisk-Users] PCI expansion slots.
Hello, Did anyone use PCI expansion slots such as: http://www.cyberresearch.com/store/product/311.2.htm I want to know how well does it work with Asterisk FXO/FXS cards? Also, does FXO/FXS drivers work automatically (meaning seemlessly recognize the expansion slots) without any Power/Bandwidth/Interrupt issues? Any alternative or information about working (or not working) baords would be highly appreciated. Thanks. ** -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Asterisk Sent: Saturday, January 31, 2004 6:03 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] 8 lines - best approach Thanks for the overwhelming response guys. Just wait for some time (6 weeks) and some of you will get to test it for sure. Watch out this mailing list for the announcement. For now, let's keep it a little secret. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jim Thompson Sent: Saturday, January 31, 2004 5:17 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] 8 lines - best approach Rob Fugina writes: On Sat, Jan 31, 2004 at 12:00:49PM -0800, Asterisk wrote: How about a 16 port FXO/FXS card (your choice of FXO/FXS modules or any mix of them) for $999. Will that be a good option? I know (me) someone (me) who'd make a (me) really good (me) beta tester (me)... I know someone else (me) who would buy them (me). -- Speed, it seems to me, provides the one genuinely modern pleasure. -- Aldous Huxley (1894 - 1963) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dial via sip gateway?
On Sat, 31 Jan 2004, Rich Adamson wrote:
I'm having a brain fart
What's the proper syntax for dialing out via a sip g/w (Mediatrix)?
Been trying stuff similar to:
exten = _6X.,1,Dial(SIP/[EMAIL PROTECTED]/${EXTEN-1})
where 3091 is alias for the port on the Mediatrix. Sniffer indicates * did
even try the IP.
should you say ${EXTEN:1} rather than ${EXTEN-1} to drop that 6 off the
front of the extension?
Greg
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Re: [Asterisk-Users] How to delay dialing
On Saturday 31 January 2004 11:23, John Baker wrote: On Sat, 2004-01-31 at 10:21, Tilghman Lesher wrote: I'll look at it again, but since I don't have a VoiceTronix card installed in any of my machines yet, I can't test it directly. I've got one I might be able to loan you for a couple of months, if you have a great desire to hack away. Thanks, but I already have a card; it simply isn't installed yet. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dial via sip gateway?
Rich Adamson wrote:
I'm having a brain fart
What's the proper syntax for dialing out via a sip g/w (Mediatrix)?
Been trying stuff similar to:
exten = _6X.,1,Dial(SIP/[EMAIL PROTECTED]/${EXTEN-1})
where 3091 is alias for the port on the Mediatrix. Sniffer indicates * did
even try the IP.
Rich
from my extensions.conf:
;**
[trunk-local]
;**
exten = _9NXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _9NXX,2,Congestion
[trunk-toll]
exten = _91NXXNXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _91NXXNXX,2,Congestion
--
Bob Knight
[-w] the work option
[EMAIL PROTECTED]
925-449-9163
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RE: [Asterisk-Users] determining legal VoIP service
In order to sell it to ITXC, the minimum capacity is 1E1/T1. We are buying/selling termination without capacity limit. If you have some good routes, please let me know Jeff Chen UM Network, Canada Tel:1-416-324-8066 Fax: 1-416-324-8261 www.mutualphone.com Yahoo messenger ID: jeffcheny2k -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: January 31, 2004 2:33 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] determining legal VoIP service -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Walker Haddock Sent: Friday, January 30, 2004 5:52 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] determining legal VoIP service Can anyone recommend who we can consult with that could provide advice on the legality of a proposed VoIP service. Specifically, we would provide VoIP termination in the USA to clients in Spain, Nigeria and Guana. The termination service would connect the VoIP clients to the PSTN through a carrier like MCI, Verizon, etc. The calls placed would connect anywhere in the world via the USA carrier. If you're interested in receiving traffic for those locations, you could talk to ITXC. They mostly sell H.323 termination though other people's POP sites around the world. They may need more termination in those areas. If so, they are very well versed in this type of thing, and could probably help you out (not just in getting your proosed plan running, but possibly making a good bit of money on top of that). I don't know anyone in the reseller/sales division, only their engineers, but you might want to give them a try. The could become a very good customer/peer of yours if you happen to be terminating in the right spots. http://www.itxc.net FYI, I don't see much on their web site about becoming a SNOC as they call it, but I'd try just giving sales or a general number a call and see who you can get transferred to. Best of luck, Daryl G. Jurbala BMPC Network Operations Tel (NY): +1 917 477 0468 x235 Tel (MI): +1 616 608 0004 x235 Tel (UK): +44 208 792 6813 x235 Fax: +1 508 526 8500 INOC-DBA: 26412*DGJ PGP Key: http://www.introspect.net/pgp ___ 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] rtp sound quality?
Rich Adamson wrote: Thanks Bob, that fixed it. Any other hints/issues/default values that I should muck with, or is that about it? Seems like it works pretty good; excellent echo cancellation, etc. I haven't done anything with the box as yet for dialing outbound. Anything to be concerned with, special parameters, etc? I can't think of anything off the top of my head. It has been a while since I set mine up. My one and only complaint so far with this box is the snmp config stuff. They only give you a windows version. I have no windows boxes in my office. I just thought some day I would have to slam together a few little snmp scripts or gui code that drives off their MIB files. But I never had to go back into the box to do anything, so this has been a low priority. I am just a low level c hack. Before I go out and write any thing to do this snmp admin stuff, are there any linux tools I could use to do this? -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ 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] OT:Linux(or *BSD) SNMP tools (Was: Re: rtp sound quality?)
On Saturday 31 January 2004 21:31, you wrote: CHOP I am just a low level c hack. Before I go out and write any thing to do this snmp admin stuff, are there any linux tools I could use to do this? Net-SNMP (http://freshmeat.net/projects/net-snmp/ , formerly UCB-SNMP or something) is very handy for this. Cheers, 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
Re: [Asterisk-Users] Dial via sip gateway?
Bob, I have a question into mediatrix for this, but maybe you have
figured it out. I am trying to map a SIP user to a specific PSTN line. I
have my extensions.conf file as you show below, but on the 1204, it just
grabs whatever line is available, whereas I want extension 101 to always
be port1 on 1204, and extension 102 to be port 2 and so on. I noticed a
NetToPstnSourceFilter MIB per port, and their docs hint at using this,
but the example in the docs describes their FXS to FXO, so I am not sure
what I would put in that MIB. CallerID info? * calling sip extension
number? Have you been able to make this work?
On Sat, 2004-01-31 at 20:22, Bob Knight wrote:
Rich Adamson wrote:
I'm having a brain fart
What's the proper syntax for dialing out via a sip g/w (Mediatrix)?
Been trying stuff similar to:
exten = _6X.,1,Dial(SIP/[EMAIL PROTECTED]/${EXTEN-1})
where 3091 is alias for the port on the Mediatrix. Sniffer indicates * did
even try the IP.
Rich
from my extensions.conf:
;**
[trunk-local]
;**
exten = _9NXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _9NXX,2,Congestion
[trunk-toll]
exten = _91NXXNXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _91NXXNXX,2,Congestion
--
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