Re: [Asterisk-Users] UPDATE - Echo cancellation, when softwaredoesn't cut it. Whats next?
No it points to Cell phone companies having better hardware echo cancellation on their lines, also cell phones themselves have a hardware echo can built in. - Original Message - From: Mike Benoit [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, July 12, 2004 1:52 AM Subject: Re: [Asterisk-Users] UPDATE - Echo cancellation, when softwaredoesn't cut it. Whats next? Here's an update on my progress for all who are interested. After carrying out many more hours of testing, the only thing that made a significant difference was changing the mainboard/CPU of my asterisk server. My original Asterisk server was a Celeron 533 with 128mb ram. Now, keep in mind, even with 5 channels in use at a time, the CPU usage was always minimal, the load never went above 0.2 that I saw. First I upgraded to a brand new Celeron 2.4Ghz with 128mb ram, and immediately noticed an improvement in the echo. Incoming calls now have virtually no echo. I have to really try to hear it. Outgoing calls still have echo, but after about 30 seconds it mostly goes away during normal conversation. Still not 100% acceptable though. However it is a huge improvement. The weird part though is this. Outgoing calls to cell phone numbers (tried 3 different ones) have virtually no echo. Outgoing calls to land line numbers do seem to have echo. I then downgraded my asterisk server to a P3-800 with 128mb ram, and I didn't notice any difference from the Celeron 2.4. So in my case just upgrading a Celeron 533 to P3-800 made a noticeable difference, but anything more than that did not. What I did notice is in /proc/interrupts, the P3-800 displays: IO-APIC-level wcfxo Whereas I believe the older Celeron 533 displayed: XT-PIC wcfxo So my guess is its not the CPU speed at all, just the way interrupts are handled. So the two questions remain. 1. Why do incoming calls have nearly no echo (sound great), and outgoing calls are bad during the first 30 seconds, and okay (but not good) after that. 2. Why do outgoing calls to cell phone numbers sound great? Seeing as an outgoing call to a land line has echo, but the same land line calling in has virtually no echo, does this point the finger at Asterisk code having issues? On Wed, 2004-06-30 at 16:36 -0700, Mike Benoit wrote: Over the last couple weeks I've tried everything I could get my hands on in an attempt to get rid of my echo problems. Using a CVS checkout of just yesterday, I've tried every echo cancellation routine in zconfig.h (including Mark2 w/Aggressive) , as well as the echotraining=800 mentioned on this list just last week. While some things worked better then others, I would consider none acceptable solutions in my situation. Playing with rx/tx gain values just seemed to quiet the voice down and along with that the echo happened to be less noticeable. I could almost get the echo to disappear with a low enough rx/tx gain, but then the voice could barely be heard, or DTMF tones stopped working. So whats the next step? I only get echo when dialing over the PSTN. Using Nufone to dial a PSTN number results in absolutely zero echo. Do I put in a request for a Telco technician to come out and take a look at the lines? One page on the Wiki says: Most of the telco's have technicians with the equipment necessary to help find the problem if the problem really is their outside plant. However, getting to that person can be a real challenge. Any suggestions on ways to overcome the challenge of getting the right technician on the phone? Thanks. -- Mike Benoit [EMAIL PROTECTED] ___ 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] New Asterisk bounty: SIP simultaneous
So isn't this the problem * has? The first client registers as the address of record, then the second client comes in with the same registration and becomes the address of record? I think you are making this look more complicated than it actually is. We do this with our SER Network all the time. Its called parallel forking. For example, our subscribers can have 2 or more Sipuras with the same number and registration info. They have one Sipura at the office and another at home. When a call is destined for that sub, SER will lookup the location database to see where it should send the INVITE. If it sees 2 or more locations then it sends multiple INVITES, ie.. Parallel Fork. The first INVITE to answer will be the one that establishes the RTP Session and all the others will receive a CANCEL. Its quite simple and works perfectly. -- Andres Network Admin http://www.telesip.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] UPDATE - Echo cancellation, when softwaredoesn't cut it. Whats next?
That doesn't explain why a incoming call from a land line has nearly no echo, while an outgoing call to the same land line has echo. Also it has always been near end echo I'm hearing, and prior to upgrading the mainboard/CPU I heard echo when calling the same cell phones. On Mon, 2004-07-12 at 02:06 -0400, Anton wrote: No it points to Cell phone companies having better hardware echo cancellation on their lines, also cell phones themselves have a hardware echo can built in. - Original Message - From: Mike Benoit [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, July 12, 2004 1:52 AM Subject: Re: [Asterisk-Users] UPDATE - Echo cancellation, when softwaredoesn't cut it. Whats next? Here's an update on my progress for all who are interested. After carrying out many more hours of testing, the only thing that made a significant difference was changing the mainboard/CPU of my asterisk server. My original Asterisk server was a Celeron 533 with 128mb ram. Now, keep in mind, even with 5 channels in use at a time, the CPU usage was always minimal, the load never went above 0.2 that I saw. First I upgraded to a brand new Celeron 2.4Ghz with 128mb ram, and immediately noticed an improvement in the echo. Incoming calls now have virtually no echo. I have to really try to hear it. Outgoing calls still have echo, but after about 30 seconds it mostly goes away during normal conversation. Still not 100% acceptable though. However it is a huge improvement. The weird part though is this. Outgoing calls to cell phone numbers (tried 3 different ones) have virtually no echo. Outgoing calls to land line numbers do seem to have echo. I then downgraded my asterisk server to a P3-800 with 128mb ram, and I didn't notice any difference from the Celeron 2.4. So in my case just upgrading a Celeron 533 to P3-800 made a noticeable difference, but anything more than that did not. What I did notice is in /proc/interrupts, the P3-800 displays: IO-APIC-level wcfxo Whereas I believe the older Celeron 533 displayed: XT-PIC wcfxo So my guess is its not the CPU speed at all, just the way interrupts are handled. So the two questions remain. 1. Why do incoming calls have nearly no echo (sound great), and outgoing calls are bad during the first 30 seconds, and okay (but not good) after that. 2. Why do outgoing calls to cell phone numbers sound great? Seeing as an outgoing call to a land line has echo, but the same land line calling in has virtually no echo, does this point the finger at Asterisk code having issues? On Wed, 2004-06-30 at 16:36 -0700, Mike Benoit wrote: Over the last couple weeks I've tried everything I could get my hands on in an attempt to get rid of my echo problems. Using a CVS checkout of just yesterday, I've tried every echo cancellation routine in zconfig.h (including Mark2 w/Aggressive) , as well as the echotraining=800 mentioned on this list just last week. While some things worked better then others, I would consider none acceptable solutions in my situation. Playing with rx/tx gain values just seemed to quiet the voice down and along with that the echo happened to be less noticeable. I could almost get the echo to disappear with a low enough rx/tx gain, but then the voice could barely be heard, or DTMF tones stopped working. So whats the next step? I only get echo when dialing over the PSTN. Using Nufone to dial a PSTN number results in absolutely zero echo. Do I put in a request for a Telco technician to come out and take a look at the lines? One page on the Wiki says: Most of the telco's have technicians with the equipment necessary to help find the problem if the problem really is their outside plant. However, getting to that person can be a real challenge. Any suggestions on ways to overcome the challenge of getting the right technician on the phone? Thanks. -- Mike Benoit [EMAIL PROTECTED] ___ 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 -- Mike Benoit [EMAIL PROTECTED] ___ 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] New Asterisk bounty: SIP simultaneous
Well Andres is right but there are numerous problems with quite a few SIP clients that do NOT follow the the SIP RFC correctly. There is a problem with dialog creation in a number of SIP products out there. SIP dialog creation is the critical part of the spec that supports parallel forking - so be careful. Jason -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andres Sent: 12 July 2004 08:54 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous So isn't this the problem * has? The first client registers as the address of record, then the second client comes in with the same registration and becomes the address of record? I think you are making this look more complicated than it actually is. We do this with our SER Network all the time. Its called parallel forking. For example, our subscribers can have 2 or more Sipuras with the same number and registration info. They have one Sipura at the office and another at home. When a call is destined for that sub, SER will lookup the location database to see where it should send the INVITE. If it sees 2 or more locations then it sends multiple INVITES, ie.. Parallel Fork. The first INVITE to answer will be the one that establishes the RTP Session and all the others will receive a CANCEL. Its quite simple and works perfectly. -- Andres Network Admin http://www.telesip.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 ___ 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] Stopping reinvite with IAX2?
Brian K. West wrote: per peer bkw Brian, What will happen to SIP UA call flow and notransfer is left at its default value?(Presumming SIP UA has canreinvite=yes) Would SIP UA stay with original server? Or? Ta SJ ___ 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] Stopping reinvite with IAX2?
Brian K. West wrote: per peer bkw - Original Message - From: Michael Graves [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, July 11, 2004 9:25 PM Subject: Re: [Asterisk-Users] Stopping reinvite with IAX2? Is this set on a per peer basis, or in the general section? Michael Actually, as a result of bug 1579, it can also be applied to the general section, if using CVS. Regards, Richard ___ 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 on FreeBSD 4.10 dies
Dr. Rich Murphey wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Arjan On Sun, 11 Jul 2004 at 15:39 -0500, Dr. Rich Murphey wrote: You might check login class in login.conf for the user that invokes asterisk. Setting cputime=unlimited may help. This will prevent the kernel from killing the process but I'm puzzled by the load Asterisk generates on a AMD XP+ 2000 cpu. While running the box goes to 40%, even though Asterisk is doing nothing (well at least: not handling calls, etc). That sounds like a bug. One should be able to attach to the process in gdb, stop the process and see where it's looping. Rich A slightly similar observation, which I assume is normal as the boxes work fine, is both my P4 2.4GHz Linux asterisks spike up to 100% load, about every 30 seconds, with no calls being handled. The boxes are only running asterisk, ntpd, sshd and the core 2.4 kernel services and the load can be observed in top by noting the system CPU usage figure in the upper part of the top display - no CPU usage is shown by any of the listed processes. As I say, it is probably normal, but I've wondered what causes it. Richard ___ 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] RE: How to differentiate a *busy* call from not available?
IsChanAvail() application might help Atif Sent via the WebMail system at convergence.com.pk ___ 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 on FreeBSD 4.10 dies
On Mon, 2004-07-12 at 02:38, Richard Scobie wrote: A slightly similar observation, which I assume is normal as the boxes work fine, is both my P4 2.4GHz Linux asterisks spike up to 100% load, about every 30 seconds, with no calls being handled. You don't mention it, but it sounds like you are running either Fedora core or Red Hat. There are known problems regarding the threading in those distos. The is a known work around too. The boxes are only running asterisk, ntpd, sshd and the core 2.4 kernel services and the load can be observed in top by noting the system CPU usage figure in the upper part of the top display - no CPU usage is shown by any of the listed processes. If you haven't done so, take a moment to read or listen about quantum mechanics. The problems with wuantum mechanics is similar to what you describe. The tool you are using to make measurements affects what you are measuring. Something to think about. Top is a very crude way of measuring system load. It is nice and useful, but remember it's short comings. As I say, it is probably normal, but I've wondered what causes it. On BSD, I think it was mentioned that select works differently and so while it is normal, it isn't what one would want. -- Steven Critchfield [EMAIL PROTECTED] ___ 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] Debian Unstable Claims Asterisk 1.0-1
Howdy, I just did an apt-get dist-upgrade on my Debian unstable box, and noticed that the Asterisk version appears to be 1.0-1 in the unstable tree. I KNOW that 1.0 hasn't been released yet, so I am wondering who is responsible for the Debian packages? This will be VERY VERY confusing for people and it should be corrected ASAP. I filed a bug into Debians bug system. ___ 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] RE: MeetMe Improvement
is there any option of inviting some one to conference, I mean, I press * for menu, then system asks me to invite some one dial 1, and then asks me to dial the extension of that person, and then call is placed to invite that person to conference. Thank you Atif Sent via the WebMail system at convergence.com.pk ___ 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 on FreeBSD 4.10 dies
Steven Critchfield wrote: On Mon, 2004-07-12 at 02:38, Richard Scobie wrote: A slightly similar observation, which I assume is normal as the boxes work fine, is both my P4 2.4GHz Linux asterisks spike up to 100% load, about every 30 seconds, with no calls being handled. You don't mention it, but it sounds like you are running either Fedora core or Red Hat. There are known problems regarding the threading in those distos. The is a known work around too. No, I have avoided the later RedHat distros for that reason. It is a stripped down RH 7.3 and updates with a custom compiled bare bones 2.4.21 kernel, (from memory - I'll be updating once .27 is out). As far as I know, threading is pretty standard there. If you haven't done so, take a moment to read or listen about quantum mechanics. The problems with wuantum mechanics is similar to what you describe. The tool you are using to make measurements affects what you are measuring. Something to think about. Top is a very crude way of measuring system load. It is nice and useful, but remember it's short comings. True, but it is out of character with what I see on similar boxes not running asterisk. What would one use to measure this less obtrusively? Regards, Richard ___ 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] New Asterisk bounty: SIP simultaneous
Paul Mahler wrote: Well, this is certainly getting exciting. Yes, it is. Sorry for coming in late to this debate... Andy, I took your advice and re-read the RFP. It's actually RFC, not RFP. (teasing :-) So, gentlemen, help me out here. The spec says: The Address of record is the SIP address that the registry knows the registrand. . . The Address of record is the public SIP uri you want people to call you at, regardless of the address of the phone you are answering on. It's the SIP phone address you place on your business card. A client uses the REGISTER method to register the address listed in the TO header field with a SIP Server. A client registers a temporary address, the address to a SIP UA, to the SIP registrar that is responsible for the domain in the AOR. This tells the SIP registrar (or location server) where to find you if someone calls your URI. When sending mail, I am not addressing the mail to the IP address you are reading the mail on, I am using your public e-mail address that is mapped to an e-mail server that is responsible for all e-mail to your domain. Later on, you fetch the e-mail from an e-mail client somewhere, with an IP address that propably changes as you travel around signing books ;-) SIP works the same way. You have a public address and a SIP proxy being responsible for keeping track of where you want to answer your calls. You can surely register several phones that you want to answer on. The proxy takes care of hiding this to the callee, so that the caller only get one set of responses. That's what the forking stuff is all about. If one phone is busy and the other one is answering, we should only signal 200 OK in SIP lingo to the caller. I don't see how two different clients can register with a server as the same address of record. Doesn't the second registration from a new client change the address of record for the registered client? You have one address-of-record that maps into several SIP URIs, one for each device. These are not as long-term as your a-o-r SIP URI. From the RFC: Location Service: A location service is used by a SIP redirect or proxy server to obtain information about a callees possible location(s). It contains a list of bindings of address-of-record keys to zero or more contact addresses. The bindings can be created and removed in many ways; this specification defines a REGISTER method that updates the bindings. If the second client is trying the same registration as the first client, and it's the responsibility of the client to provide the complete list of bindings, how does the second client know the list of bindings for the first client that bound the registration? It's *not* the responsibility of the *client* to provide a list, it's the server that responds with a list, telling the client by the way, these devices are also registred for the same a-o-r. So isn't this the problem * has? The first client registers as the address of record, then the second client comes in with the same registration and becomes the address of record? The address of record does not change because of a registration. The stored address (the contact: header) of where we can reach you (location) changes. And yes, if you have multiple devices registering for the same Asterisk sip [peer] account, it will be changing for each registration. This is not the behaviour of most SIP Proxys. Asterisk is *not* a SIP proxy. It's a SIP registrar and location server. It's a very clever SIP UA. It wants to be in the middle of the call and wants to be in control of each device. This device-slave view doesn't match the SIP architecture. Due to Asterisk's multi-protocol architecture we have to make some compromises in the SIP channel to be able to have some kind of generic view of calls and phones in the core. A SIP proxy is never the end point of a call, it should never handle the media stream. The power is in the edge, in the phones. This is why transfers and other PBX functions is a bit messy with SIP and Asterisk, we are trying to find a way to do it centralized as Asterisk but de- centralized as SIP... I've spent a considerable amount of time investigating support for multiple registrations on one Asterisk sip [peer] account and after learning about Asterisk's architecture come to the conclusion that it is not an easy or even a desirable feature to implement. The architecture of Asterisk is a PBX, and the dial plan and a lot of apps wants to be in control of the device. It may be possible, but will probably lead to a lot of changes to Asterisk, both core and applications, that no other channel will benefit from. A quick hack to support it may lead to a lot of confusion on how to handle other apps. And it's a lot more work than the bounty will cover. I suggest that you use a forking SIP proxy in conjunction with Asterisk to get this functionality. If you are looking for a SIP PBX, check Pingtel's Open Source software. If you are looking for a SIP proxy, test SIP Express Router from
[Asterisk-Users] Problem with character encoding in SIP channel (ISO vs. UTF-8)
Hi I recently noticed that asterisk passes Caller IDs and SendText messages containing sepcial characters (such as the german umlaut characters äöü) with ISO-8859-1 encoding to the SIP phone. Hence user names and text strings like Müller are not correctly displayed on the receiving phone. According to RFC 3261 SIP uses UTF-8 encoding. Shouldn't asterisk convert these characters from ISO-8859-1 to UTF-8 before passing them to SIP devices? Best regards martin -- Martin A. Blatter | lic. oec. publ. Wirtschaftsinformatiker | IT-Leiter OLMeRO AG | Europastrasse 30 | CH-8152 Glattbrugg blatter-at-olmero.ch | IAXtel 1-700-200-4450 ___ 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] New Asterisk bounty: SIP simultaneous
* No, there's no quick fix for a 100 USD bounty How much you estimate on quick fix? -Kannaiyan. ___ 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] feature - VM gain adjust?
Hello I'm toying with adding a feature request to provide some sort of gain setting for voicemail when accessed from certain interfaces. Maybe something like voicemail=6.0 (db) within a specific channel section of zapata.conf corresponding to a pstn line. That gets my vote. We experience this low-volume voicemail problem. (and I spent a long time looking for the proposed setting to tweak!) Bob ___ 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] New Asterisk bounty: SIP simultaneous
Excellent Post! Very Informative. Thanks a lot Sir! Regards, Girish From: Olle E. Johansson [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous Date: Mon, 12 Jul 2004 10:52:33 +0200 Paul Mahler wrote: Well, this is certainly getting exciting. Yes, it is. Sorry for coming in late to this debate... Andy, I took your advice and re-read the RFP. It's actually RFC, not RFP. (teasing :-) So, gentlemen, help me out here. The spec says: The Address of record is the SIP address that the registry knows the registrand. . . The Address of record is the public SIP uri you want people to call you at, regardless of the address of the phone you are answering on. It's the SIP phone address you place on your business card. A client uses the REGISTER method to register the address listed in the TO header field with a SIP Server. A client registers a temporary address, the address to a SIP UA, to the SIP registrar that is responsible for the domain in the AOR. This tells the SIP registrar (or location server) where to find you if someone calls your URI. When sending mail, I am not addressing the mail to the IP address you are reading the mail on, I am using your public e-mail address that is mapped to an e-mail server that is responsible for all e-mail to your domain. Later on, you fetch the e-mail from an e-mail client somewhere, with an IP address that propably changes as you travel around signing books ;-) SIP works the same way. You have a public address and a SIP proxy being responsible for keeping track of where you want to answer your calls. You can surely register several phones that you want to answer on. The proxy takes care of hiding this to the callee, so that the caller only get one set of responses. That's what the forking stuff is all about. If one phone is busy and the other one is answering, we should only signal 200 OK in SIP lingo to the caller. I don't see how two different clients can register with a server as the same address of record. Doesn't the second registration from a new client change the address of record for the registered client? You have one address-of-record that maps into several SIP URIs, one for each device. These are not as long-term as your a-o-r SIP URI. From the RFC: Location Service: A location service is used by a SIP redirect or proxy server to obtain information about a callees possible location(s). It contains a list of bindings of address-of-record keys to zero or more contact addresses. The bindings can be created and removed in many ways; this specification defines a REGISTER method that updates the bindings. If the second client is trying the same registration as the first client, and it's the responsibility of the client to provide the complete list of bindings, how does the second client know the list of bindings for the first client that bound the registration? It's *not* the responsibility of the *client* to provide a list, it's the server that responds with a list, telling the client by the way, these devices are also registred for the same a-o-r. So isn't this the problem * has? The first client registers as the address of record, then the second client comes in with the same registration and becomes the address of record? The address of record does not change because of a registration. The stored address (the contact: header) of where we can reach you (location) changes. And yes, if you have multiple devices registering for the same Asterisk sip [peer] account, it will be changing for each registration. This is not the behaviour of most SIP Proxys. Asterisk is *not* a SIP proxy. It's a SIP registrar and location server. It's a very clever SIP UA. It wants to be in the middle of the call and wants to be in control of each device. This device-slave view doesn't match the SIP architecture. Due to Asterisk's multi-protocol architecture we have to make some compromises in the SIP channel to be able to have some kind of generic view of calls and phones in the core. A SIP proxy is never the end point of a call, it should never handle the media stream. The power is in the edge, in the phones. This is why transfers and other PBX functions is a bit messy with SIP and Asterisk, we are trying to find a way to do it centralized as Asterisk but de- centralized as SIP... I've spent a considerable amount of time investigating support for multiple registrations on one Asterisk sip [peer] account and after learning about Asterisk's architecture come to the conclusion that it is not an easy or even a desirable feature to implement. The architecture of Asterisk is a PBX, and the dial plan and a lot of apps wants to be in control of the device. It may be possible, but will probably lead to a lot of changes to Asterisk, both core and applications, that no other channel will benefit from. A quick hack to support it may lead to a lot of confusion on how to handle other apps. And it's a lot more work than the bounty will cover. I
RE: [Asterisk-Users] QoS in asterisk
On 11 Jul 2004 at 19:16, Rich Adamson wrote: QoS is most certainly an issue when making the decision to move off the PSTN. Is the performance of your VoIP system going to be comparable to the performance of your PSTN system? Sounds like a reasonable question to me. Not trying to get in the middle of whatever argument you're trying to make, the poster's original question (although probably not worded all that clear) can be answered by... no, asterisk cannot make a decision to route calls via a second path due to quality issues on some first choice path. Well...you could run an agi to check ping time for 1 sec and then if the differences are too much or the overall amount is too high, then use the POTS line... Matt Riddell ___ 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] New Asterisk bounty: SIP simultaneous
Kannaiyan Natesan wrote: * No, there's no quick fix for a 100 USD bounty How much you estimate on quick fix? I apologize for my Swenglish language... I don't believe there's a quick fix at all. If you want a quote for a fix, contact me off-list. But remember, that I believe that fixing this is chan_sip *will* cause confusion and errors to happen in other parts of Asterisk. In order to provide a better answer, I need some time and funding to research this a bit more. Every problem has a solution. /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] feature - VM gain adjust?
That gets my vote. We experience this low-volume voicemail problem. (and I spent a long time looking for the proposed setting to tweak!) Think about a dynamic sound compressor that would possibly auto-adjust. ___ 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] UPDATE - Echo cancellation, when software doesn't cut it. Whats next?
So the two questions remain. 1. Why do incoming calls have nearly no echo (sound great), and outgoing calls are bad during the first 30 seconds, and okay (but not good) after that. 2. Why do outgoing calls to cell phone numbers sound great? Seeing as an outgoing call to a land line has echo, but the same land line calling in has virtually no echo, does this point the finger at Asterisk code having issues? Echo (most often) comes from hybrid circuits on PSTN lines (2 wires - 4 wires transformation). Cell phones, as well as some corporate digital phones don't go through that kind of devices, so there is no echo generated. So, basically, no echo cancellation required. Unfortunatly, it's impossible to know from the caller point of view whether the call will need echo cancellation or not. -- Nicolas Bougues Axialys Interactive ___ 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] feature - VM gain adjust?
Hello That gets my vote. We experience this low-volume voicemail problem. (and I spent a long time looking for the proposed setting to tweak!) Think about a dynamic sound compressor that would possibly auto-adjust. Are you suggesting such a thing exists, or that that would be a proposed future application? Bob ___ 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] X101P FXO with RED alarm
On Sun, 11 Jul 2004 23:02:56 +0100, Richard Airlie [EMAIL PROTECTED] wrote: On Sat, Jul 10, 2004 at 05:55:21PM +0100, Kevin Walsh wrote: Richard Airlie [EMAIL PROTECTED] wrote: First things first. Scrap the ports and build from the latest CVS source. 0.9 is far to old and buggy, and suspect the same of the Zaptel driver you have, although I don't use *BSD myself. I cvsup'd to the latest source yesterday and tried to build zaptel, but it failed right away. (trying to include linux/*.h) You need to get zaptel built correctly with your kernel otherwise it will never run correctly. Jason ___ 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] permission problem
Hi everybody, Is the only way to use asterisk _not_ as root to change the permission of all the directories where asterisk need to create a file? (/var/run/, /var/log/asterisk/messages) any help will be appreciated, Cyprien ___ 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] Can I hear voice messages from diax phone button directly ?
Hi, I'm testind Diax. I have flashing note about 1 new voice message. Can I hear it somehow from Diax gui, or must I call pbx to get message ? Thanks, 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] New Asterisk bounty: SIP simultaneous
On 11/07/2004 at 18:11 Paul Mahler wrote: Well, this is certainly getting exciting. Andy, I took your advice and re-read the RFP. Andy--I don't think you are a Sorry, I was sleeping when these new emails came in I've read the other responses which seem to make it pretty clear.. and address all the points and give most of the info you need...(do I need to add to it?) I couldn't for the life of me remember the name (it was late) and Andres reminded us all that it's called Parallel Forking - it's by far the best feature of SIP and nearly, but not quite, negates the NAT problems. The reason i've been so adamant about this, is that I use it every day... my * box and 2 of my phones register with a local sip proxy for the same sip address... I use this just incase my * box dies, since it's my development box too and I'm always mesing with it. good candidate for a beginner's book on *, but if you send my your address, I'll send you a copy on me. :-) Or some Ninja assasin... ;) Perhaps you could also sign it :D (not the Ninja assasin ;) ) Andy, I'm in your hands. I was too late... I took the liberty of getting some sleep... appologies. Andy ___ 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] Can I hear voice messages from diax phone button directly ?
Hi Robert, - Original Message - From: Robert Rozman [EMAIL PROTECTED] I'm testind Diax. I have flashing note about 1 new voice message. Can I hear it somehow from Diax gui, or must I call pbx to get message ? You need to call Asterisk to get the message. Diax just gives you the number of new/old messages available in your voice mailbox. You can eventually define a speed dial for the voicemail and then enter directly based on your caller ID, without asking for mailbox number and password. Best regards, Dan ___ 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] permission problem
I modified the permissions of /var/spool/asterisk and /var/log/asterisk and it seems that asterisk is launching now. But I still have messages at the beginning telling me that: Unable to open pid file '/var/run/asterisk.pid': Permission denied Unable to bind socket to /var/run/asterisk.ctl: Address already in use Any ideas if it's bad, or if I can just forgot about it? Cyprien On Monday 12 July 2004 13:18, Cyprien Simons wrote: Hi everybody, Is the only way to use asterisk _not_ as root to change the permission of all the directories where asterisk need to create a file? (/var/run/, /var/log/asterisk/messages) any help will be appreciated, Cyprien ___ 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] feature - VM gain adjust?
Are you suggesting such a thing exists, or that that would be a proposed future application? I propose to think if an AGC / dynamic compressor could be used instead of a config variable. Most sound editors have modules for this. ___ 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] E100P and T1 channel banks
Could you kind Asterians (should we pick Asteroids then?) confirm if I can use an E100P card with a T1 channel bank via * please? I live in the UK hence the question. Luan One UK Asteroid (...this sounds better I think) ___ 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] New Asterisk bounty: SIP simultaneous
I don't think we should let these misunderstandings judge the quality of Paul's Asterisk book. Even authors need to learn now and then :-) Can I just point out that the reason I said what I said (see, I can't write) was because Paul steadfastly refused to believe what we were saying, rather than investigating it.ie His response was more like: You're wrong, I'm right. rather than: Oh... maybe there's something I'm not aware of. I shall investigate immediately. I'll admit techies always argue over stuff like this, primarily because they don't want to be seen to not know something.. anyho.. I'd consider the discussion of the existance of forking closed and proven and we can now begin arguing over why it would/wouldn't be a good idea to include this behaviour in * ;) Andy ___ 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] feature - VM gain adjust?
Are you suggesting such a thing exists, or that that would be a proposed future application? I propose to think if an AGC / dynamic compressor could be used instead of a config variable. Most sound editors have modules for this. So how would you detect the remote caller is 14.7 db away from * and adjust the 'outbound' voice message to be at some higher audio level? I like the AGC approach, but I'm not sure its realistic in terms of consistently being able to identify the transmission loss from each and every vm call. Since we know what the loss is for each pstn line (to the central office), it would appear that static value would be a good starting point and the user could adjust from there. Much easier (and more likely) to implement. ___ 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] QoS in asterisk
how do you ping a TDM connection ? On Mon, 2004-07-12 at 11:43, [EMAIL PROTECTED] wrote: On 11 Jul 2004 at 19:16, Rich Adamson wrote: QoS is most certainly an issue when making the decision to move off the PSTN. Is the performance of your VoIP system going to be comparable to the performance of your PSTN system? Sounds like a reasonable question to me. Not trying to get in the middle of whatever argument you're trying to make, the poster's original question (although probably not worded all that clear) can be answered by... no, asterisk cannot make a decision to route calls via a second path due to quality issues on some first choice path. Well...you could run an agi to check ping time for 1 sec and then if the differences are too much or the overall amount is too high, then use the POTS line... Matt Riddell ___ 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] E100P and T1 channel banks
On Monday 12 July 2004 07:36, luan au wrote: Could you kind Asterians (should we pick Asteroids then?) confirm if I can use an E100P card with a T1 channel bank via * please? I live in the UK hence the question. Yes. You''l only get 24 channels but it shoudl work fine. And I prefer the term Astericians (think electrician), myself. -A. ___ 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] QoS in asterisk
On Monday 12 July 2004 05:43, [EMAIL PROTECTED] wrote: Not trying to get in the middle of whatever argument you're trying to make, the poster's original question (although probably not worded all that clear) can be answered by... no, asterisk cannot make a decision to route calls via a second path due to quality issues on some first choice path. Well...you could run an agi to check ping time for 1 sec and then if the differences are too much or the overall amount is too high, then use the POTS line... Why not just work with qualify? If the connection is too lagged * won't make the call through it (although if the link BECOMES laggy it will continue to use the connection). -A. ___ 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] gnophone and asterisk
Dear All, I just do cvsup for asterisk (7/12/2004),and yesterday cvs with the same result. I'm trying to make gnophone work with asterisk. Following the wiki pages, here's my iax.conf [general] port=5036 ;bindaddr=192.168.1.145 iaxcompat=yes delayreject=yes bandwidth=low ; ;allow=all ; same as bandwidth=high ;disallow=g723.1; Hm... Proprietary, don't use it... disallow=lpc10 ; Icky sound quality... Mr. Roboto. ;allow=gsm ; Always allow GSM, it's cool :) ; [gnophone];This is the name of the user, and the reference in exten$ type=friend ;Asterisk send calls to a peer, receives calls from a $ secret=testing1 ;The is the secret in gnophone. auth=plaintext;Asterisk to Asterisk can use md5 and rsa, I do not i$ host=dynamic ;This allows the host to come from different context=sip ;What context to jump to in the extensions.conf file. $ mailbox=101 ;Which mailbox to use. callerid=Isianto 123456 ;Caller ID to show when the call is incoming from g$ permit=0.0.0.0/0.0.0.0 ;Which IP's can be incoming. here's my gnophone config mode=2 iaxserver=192.168.1.2 iaxcontext=sip iaxusername=gnophone iaxpassword=testing1 iaxpeer=gnophone iaxsecret=testing1 iaxprefix= iaxport=5036 the problem is when I start *, I can see in the console like this: Jul 12 14:43:05 WARNING[16384]: chan_iax2.c:6537 set_config: Ignoring port for now == Using TOS bits 16 == IAX Ready and Listening on 0.0.0.0 port 4569 == Loaded firmware 'iaxy.bin' -- Loaded provisioning template 'default' and then I do nmap -sU ip (I don't see port 4569 or 5036 available). I can't register gnophone with *, when I do ethereal, I can see that gnophone tried to connect to port 5036, but the * replied destination unreachable. Is there something wrong with my config? Thanks ___ 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] QoS in asterisk
On 12 Jul 2004 at 14:06, Michael Bielicki wrote: how do you ping a TDM connection ? Sorry, where does it say this is regarding a TDM connection? I use IAX trunking and a ping script to check times and fluctuations to my remote offices. Matt Riddell On Mon, 2004-07-12 at 11:43, [EMAIL PROTECTED] wrote: On 11 Jul 2004 at 19:16, Rich Adamson wrote: QoS is most certainly an issue when making the decision to move off the PSTN. Is the performance of your VoIP system going to be comparable to the performance of your PSTN system? Sounds like a reasonable question to me. Not trying to get in the middle of whatever argument you're trying to make, the poster's original question (although probably not worded all that clear) can be answered by... no, asterisk cannot make a decision to route calls via a second path due to quality issues on some first choice path. Well...you could run an agi to check ping time for 1 sec and then if the differences are too much or the overall amount is too high, then use the POTS line... ___ 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] PRI numbering plan
Hello! I have an E100P connected to our partner's PBX. They want the following: Called number must have numbering plan/type set as: unknown/unknown and calling number in: ISDN/national. I searched for the config file, but I found only pridialplan option on zaptel.conf. When I set it to unknown, the called number has unknown/unknown, however the calling number has as well. When I set national, the calling number is ISDN/national but the called number is national as well - so I can't establish connection. Is it possible to set those numbering plans/types differently for called and calling number? In other case I can't place call or won't see the calling number on the phone. Thanks in advance, Thomas ___ 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] QoS in asterisk
On 12 Jul 2004 at 8:22, Andrew Kohlsmith wrote: On Monday 12 July 2004 05:43, [EMAIL PROTECTED] wrote: Not trying to get in the middle of whatever argument you're trying to make, the poster's original question (although probably not worded all that clear) can be answered by... no, asterisk cannot make a decision to route calls via a second path due to quality issues on some first choice path. Well...you could run an agi to check ping time for 1 sec and then if the differences are too much or the overall amount is too high, then use the POTS line... Why not just work with qualify? If the connection is too lagged * won't make the call through it (although if the link BECOMES laggy it will continue to use the connection). Qualify will only stop the call going through if for example the ping is above 200ms. I find most of my problems come from fluctuating ping times (~100ms) than from a stable high ping. Matt Riddell ___ 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] QoS in asterisk
Doesn't make any difference 'how' one might ping a remote site, ping will never qualify the Quality of the channel between two points. It will only suggest its up/down and possibly the delay at that specific point in time. Has nothing to do with whether packets were dropped or delayed some milliseconds before or after the ping, and the ping pkt would never be subjected to any positive QoS parameters implemented in the point-to-point network infrastructure. A large number of ISP's block icmp pkts anyway (for other reasons), so its not a reasonable way to determine anything. how do you ping a TDM connection ? On Mon, 2004-07-12 at 11:43, [EMAIL PROTECTED] wrote: On 11 Jul 2004 at 19:16, Rich Adamson wrote: QoS is most certainly an issue when making the decision to move off the PSTN. Is the performance of your VoIP system going to be comparable to the performance of your PSTN system? Sounds like a reasonable question to me. Not trying to get in the middle of whatever argument you're trying to make, the poster's original question (although probably not worded all that clear) can be answered by... no, asterisk cannot make a decision to route calls via a second path due to quality issues on some first choice path. Well...you could run an agi to check ping time for 1 sec and then if the differences are too much or the overall amount is too high, then use the POTS line... ___ 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] E100P and T1 channel banks
Andrew Kohlsmith wrote: On Monday 12 July 2004 07:36, luan au wrote: Could you kind Asterians (should we pick Asteroids then?) confirm if I can use an E100P card with a T1 channel bank via * please? I live in the UK hence the question. Yes. You''l only get 24 channels but it shoudl work fine. And I prefer the term Astericians (think electrician), myself. -A. Any signaling and framing issues? ___ 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] How to make * don't strip the leading 0
Hi folks! Is it possible to tell asterisk not to strip the leading 0 of *incoming* MSNs? I use asterisk with i4l and whenever I get a call from an long-distance party, the leading 0, which should be there according the german numbering, is not. So if I get a call from a mobile phone 0177-1234567 should be displayed, but 177-1234567 is displayed. I double checked if I've forgotten to remove an option to strip the first digit of incoming calls and found nothing. The wiki and the mailinglist archives can't enlight me either, why asterisk behaves like this, or how I can turn it off. So if someone could give me a hint, I would be very delighted! Best regards Kai -- Kai Militzer WESTEND GmbH | Internet-Business-Provider Technik CISCO Systems Partner - Authorized Reseller Lütticher Straße 10 Tel 0241/701333-11 [EMAIL PROTECTED] D-52064 Aachen Fax 0241/911879 ___ 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] PRI numbering plan
pridialplan=unknown prilocaldialplan=national Thomas wrote: Hello! I have an E100P connected to our partner's PBX. They want the following: Called number must have numbering plan/type set as: unknown/unknown and calling number in: ISDN/national. I searched for the config file, but I found only pridialplan option on zaptel.conf. When I set it to unknown, the called number has unknown/unknown, however the calling number has as well. When I set national, the calling number is ISDN/national but the called number is national as well - so I can't establish connection. Is it possible to set those numbering plans/types differently for called and calling number? In other case I can't place call or won't see the calling number on the phone. Thanks in advance, Thomas ___ 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] QoS in asterisk
Would you consider posting this this to the wiki? :) I think that would be great. On Mon, 2004-07-12 at 08:35, [EMAIL PROTECTED] wrote: On 12 Jul 2004 at 14:06, Michael Bielicki wrote: how do you ping a TDM connection ? Sorry, where does it say this is regarding a TDM connection? I use IAX trunking and a ping script to check times and fluctuations to my remote offices. Matt Riddell On Mon, 2004-07-12 at 11:43, [EMAIL PROTECTED] wrote: On 11 Jul 2004 at 19:16, Rich Adamson wrote: QoS is most certainly an issue when making the decision to move off the PSTN. Is the performance of your VoIP system going to be comparable to the performance of your PSTN system? Sounds like a reasonable question to me. Not trying to get in the middle of whatever argument you're trying to make, the poster's original question (although probably not worded all that clear) can be answered by... no, asterisk cannot make a decision to route calls via a second path due to quality issues on some first choice path. Well...you could run an agi to check ping time for 1 sec and then if the differences are too much or the overall amount is too high, then use the POTS line... ___ 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 -- respectfully, Joseph - (606) 477-2355 x140 --= ___ 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] Re: gnophone and asterisk
On Mon, Jul 12, 2004 at 03:30:24PM +0700, Isianto Istiadi wrote: and then I do nmap -sU ip (I don't see port 4569 or 5036 available). I can't register gnophone with *, when I do ethereal, I can see that gnophone tried to connect to port 5036, but the * replied destination unreachable. Is there something wrong with my config? gnophone 0.2.4 uses iax only not iax2. -- Stefan Tichy [EMAIL PROTECTED] ___ 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] Re: permission problem
On Mon, Jul 12, 2004 at 01:32:39PM +0200, Cyprien Simons wrote: I modified the permissions of /var/spool/asterisk and /var/log/asterisk and it seems that asterisk is launching now. But I still have messages at the beginning telling me that: Unable to open pid file '/var/run/asterisk.pid': Permission denied Unable to bind socket to /var/run/asterisk.ctl: Address already in use Create a directory /var/run/asterisk/, change its owner to asterisk (the non-root user) and set astrundir = /var/run/asterisk in /etc/asterisk/asterisk.conf. -- Stefan Tichy [EMAIL PROTECTED] ___ 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 on FreeBSD 4.10 dies
Differences in how poll() works is probably responsible.
Try this and see if it helps.
Cheers,
Rich
-Original Message-
[mailto:[EMAIL PROTECTED] On Behalf Of Arjan
On Sun, 11 Jul 2004 at 16:03 -0500, Dr. Rich Murphey wrote:
That sounds like a bug. One should be able to attach to
the process
in gdb, stop the process and see where it's looping.
I'm going to pretend here and now that I'm a hardcore
debugger (with a little help of my local debug-guru haha):
--- ktrace output coming up -
44392 asterisk RET read 0
44392 asterisk CALL poll(0xbfa76fb4,0x1,0)
44392 asterisk RET poll 1
44392 asterisk CALL read(0x12,0x284fd0a0,0x100)
44392 asterisk GIO fd 18 read 0 bytes
root asterisk 44392 18 /var 22833 prwx-- 0 r
---
[EMAIL PROTECTED] find /var -inum 22833
/var/run/autodial.ctl
A backtrace in gdb comes up with:
#0 0x28142d64 in __sys_read () from /usr/lib/libc_r.so.4
#1 0x2813f1a0 in _read () from /usr/lib/libc_r.so.4
#2 0x2813f1fa in read () from /usr/lib/libc_r.so.4
#3 0x284fafaf in autodial (ignore=0x0) at pbx_wilcalu.c:83
#4 0x28105240 in _thread_start () from /usr/lib/libc_r.so.4
#5 0x0 in ?? ()
Does this make any sense ?
arjan
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Index: pbx_wilcalu.c
===
RCS file: /usr/cvsroot/asterisk/pbx/pbx_wilcalu.c,v
retrieving revision 1.14
diff -u -r1.14 pbx_wilcalu.c
--- pbx_wilcalu.c 22 Jun 2004 18:49:00 - 1.14
+++ pbx_wilcalu.c 12 Jul 2004 13:09:24 -
@@ -74,11 +74,18 @@
while(1){
ssize_t bytes;
void *pass;
+int ret = 0;
memset(buf,0,257);
fds[0].fd = fd;
fds[0].events = POLLIN;
- poll(fds, 1, -1);
+ ret = poll(fds, 1, -1);
+if ((ret 0)
+( (fds[0].revents == POLLHUP) ||
+ (fds[0].revents == POLLHUP))) {
+ ast_log(LOG_ERROR, Autodial: cannot poll dial file: %s\n,
dialfile);
+pthread_exit(NULL);
+}
bytes=read(fd,buf,256);
buf[(int)bytes]=0;
R: [Asterisk-Users] How to make * don't strip the leading 0
Is it possible to tell asterisk not to strip the leading 0
of *incoming* MSNs? I use asterisk with i4l and whenever
I get a call from an long-distance party, the leading 0, which
should be there according the german numbering, is not.
Are you *really* sure that the 0 is transmitted in the CLI, and that it isn't stripped
already by the phone company? I think the easiest thing for you would be to add the
leading 0 before forwarding the call to your SIP client (ie.
SetCallerID(0${CALLERIDNUM}) in your extensions.conf for each extesion where you'd
like to add the 0).
Regards
Manuel
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Re: [Asterisk-Users] How to make * don't strip the leading 0
Kai Militzer schrieb: ... Is it possible to tell asterisk not to strip the leading 0 of *incoming* MSNs? I use asterisk with i4l and whenever I get a call from an long-distance party, the leading 0, which should be there according the german numbering, is not. So if I get a call from a mobile phone 0177-1234567 should be displayed, but 177-1234567 is displayed. I double Hi, that's rather your ISDN equipment than asterisk, who strips the leading 0. (National numbering scheme) Look at isdnrep! Probably you'll find the same numbers without leading 0 there. I4L forwards those 0-less numbers to asterisk, and asterisk takes them as they are. chap_capi for my AVM Fritz card does display numbers with leading 0. I have currently the same problem with my E1 card and I wonder, how I can get asterisk to append a leading 0 before forwarding the call, for my IP phones show the correct callee number with leading 0. Roger. ___ 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 on FreeBSD 4.10 dies
Interestingly you do not get the same problem of FreeBSD 5.2.1. Chris On Sun, 2004-07-11 at 23:55, Jean-Yves Avenard wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hello On 12/07/2004, at 4:24 AM, Arjan wrote: 43676 root63 0 10244K 7628K RUN 2:44 99.05% 99.02% asterisk This is covered in the asterisk FreeBSD section: http://www.voip-info.org/tiki-index.php?page=Asterisk+FreeBSD extract: CPU 99.9 % used by Asterisk? The current version runs amok on a FreeBSD system, occuping all your CPU cycles. To get Asterisk back to a normal level, you have to disable the problemativ module in Asterisk config modules.conf with this statement: noload = pbx_wilcalu.so In any case, I gave up using Asterisk with FreeBSD too many issues that couldn't be explained. Switching to linux fixed all the issues with the exact same configuration file Jean-Yves - --- Jean-Yves Avenard Hydrix Pty Ltd - Embedding the net www.hydrix.com | fax +61 3 95093717 | phone +61 3 9509 3724 -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (Darwin) iD8DBQFA8cVmXeDVKqIr3GURAiDBAJ4yLySDKD8NoozveF8eIHD+jRWtuACeIf1M DyckWYJeN9rpjbfvxGZzQMk= =O3eG -END PGP SIGNATURE- ___ 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] How to make * don't strip the leading 0
Kai Militzer wrote: Hi folks! Is it possible to tell asterisk not to strip the leading 0 of *incoming* MSNs? I use asterisk with i4l and whenever I get a call from an long-distance party, the leading 0, which should be there according the german numbering, is not. So if I get a call from a mobile phone 0177-1234567 should be displayed, but 177-1234567 is displayed. I double checked if I've forgotten to remove an option to strip the first digit of incoming calls and found nothing. The wiki and the mailinglist archives can't enlight me either, why asterisk behaves like this, or how I can turn it off. So if someone could give me a hint, I would be very delighted! Best regards Kai Coud it be that your provider is striping 0? ___ 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] E1 config help and guidance
Darren, Many thanks for your help - I've got further, but am still stumped. Have a look at the following table: LED | ISDN| Asterisk --+---+- OOS | Out | Red ACT | Green | Green RED | Out | Red YEL | Out | Out LBK | Out | Out CC| Out | Out DCH | Green | Green The LED indicators are as follows: OOS: Out of Service ACT: Active State RED: Red alarm state detected (Could be Loss of carrier, loss of frame or loss of crc multiframe YEL: Yellow alarm state detected (remote alarm indication from remote end) LBK: Loopback mode CC: Clock controller not equipped DCH: DCH is established. What this table indicates is that if I plug the 2MB pri card from the nortel into the EuroISDN bearer box, the leds light up as shown in the ISDN column. If I plug the nortel card into the asterisk box, then the leds light up as shown in the Asterisk column. I would have expected the leds to light up in the same manner - am I missing something ? Apart from asterisk experience :) Julian -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Storer, Darren Sent: 09 July 2004 21:06 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] E1 config help and guidance Hi Julian, J I want to put asterisk in the middle of our current pbx (Meridian J Option11) Something like this?: - | | PSTN ---span1--| CPE Asterisk NET |--span2--- Nortel | | | | - Assuming that you connect your incoming Telco PRI (PSTN) to span1 and the Nortel PBX to span2 (as depicted above) the lines below should help: Extract from zaptel.conf span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 span=2,0,0,ccs,hdb3,crc4 bchan=32-46,48-62 dchan=47 Extract from zapata.conf pridialplan=local switchtype = euroisdn signalling = pri_cpe group = 1 channel = 1-10 switchtype = euroisdn signalling = pri_cpe group = 2 channel = 32-41 switchtype = euroisdn signalling = pri_net In the config lines above, span1 is set to take timing from the PSTN whilst span2 is configured to give timing to the Nortel. Span1 will behave like a piece of CPE (PBX) and span2 will behave like the NETwork. NB. The channels in group 1 and 2 are depleted as you only have 10 channels enabled on your PRI. After you have implemented the changes above (or any subsequent changes to the low level PRI config) you should, at the very least, remember to restart the Asterisk system or, as Critch advises, power down and up again. HTH Darren -- Comgate TelcoInternetBroadcast -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of asterisk Sent: 09 July 2004 19:00 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] E1 config help and guidance I've googled / voip-info'd / searched until my eyes are blurry, but couldn't see the info I was looking for. I've turned here for help! Asterisk CVS head (9/7/04) Fedora Core 2 (updated to 2.6.6 kernel) DE405P (jumpers set to E1) I want to put asterisk in the middle of our current pbx (Meridian Option11) Currently the meridian has a 2MB pri EuroISDN card linked via a rj-45 into a euroISDN bearer. This bearer only has 10 channels activated (out of the 30). Obviously, this works - handsets make external calls. What I wanted to do was to add * to the mix, in the middle so that it can intercept inbound / outbound calls and do what it needs to do, as well as providing all the extra functionality that this wonderful product provides. In order to achieve this, I assumed that I needed to take rj45 from the bearer box and plug that into span 2, and take a cable from span 1 into the bearer box. My problem (and blurry eyes) come from not understanding the various protocols to assign to each span. I want the meridian to think that it's still plugged into the EuroISDN bearer. So span 2 should be set up as a EuroISDN link ? What should span 1 be set up as ? What channels should be configured ? Any guidance (I'm not looking for the solution (would be nice!) but for pointers in the right direction). I have previously been able to set up asterisk using the x100p and graduated to BRI isdn. I just got the 405 today and wanted to play! Thanks in advance. Julian. ___ 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] using asterisk voicemail with a class 5 softswitch
Hi, which IP Centrex setup are you using? Gary I am using asterisk as a voicemail server for our IP Centrex SoftPBX. Umar. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Chad Whitten Sent: 09 July 2004 22:46 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] using asterisk voicemail with a class 5 softswitch when you say you have integration what exactly do you mean? are you using asterisk as the voicemail system for a class 5 switch? On Friday 09 July 2004 15:45, usedcanon wrote: I have integration. Asterisk is upto the task however you may need to do some work arounds. Umar. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Chad Whitten Sent: 09 July 2004 20:51 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] using asterisk voicemail with a class 5 softswitch anyone have any idea on the compatibility of asterisk voicemail with a class 5 switch that can do SIP (in particular the MetaSwitch VP3500)? -- Chad Whitten Network/Systems Administrator [EMAIL PROTECTED] 601-944-4801 Phone ___ 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 -- Chad Whitten Network/Systems Administrator [EMAIL PROTECTED] 601-944-4801 Phone ___ 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] Using MD5 to encrpty PIN
TrTryingo get * to register to a service that uses account and pin but the PIN must be encrypted using MD5. The service does not require the phone number to register to the SIP Proxy. I can get the REGISTER message to send the account by using the below register line in the [general] section of the sip.coconf register=123456789012:[EMAIL PROTECTED]/123456789012 but am still perplexed on how to send the PIN encrypted using MD5. acct = 123456789012 PIN = 1000 My [contexts], under sip.coconflooks as follows: [general] port = 5060 ; Port to bind to bibindaddr 0.0.0.0 ; Address to bind SIP channel to context = voice-mail; Default context for incoming calls dtdtmfmodefrfc33 [EMAIL PROTECTED]/17135551212 [17135551212] type=peer context=vovoicelinedtdtmfmodefrfc33 secret=1000 qualify=1000 ususername23456789012 host=dynamic dedefaultip92.168.0.1 auauthD5 ;acaccountcode23456789012 Below is the REGISTER packet sent by the InInnoMediaevice that successfully registered with the SIP proxy. SIP Header Message Type = Request Method = REGISTER Request URI = sip:192.168.0.1:5060 SIP Version = SIP/2.0 Via = SIP/2.0/UDUDP92.168.0.49:5060;branch=z9hGhGKbKd576315060-11 (Path Taken By Request Till Now) From = sip:[EMAIL PROTECTED]:5060;tag=D835763113C4-002120DC0 (Request Initiator) To = sip:[EMAIL PROTECTED]:5060 (Recipient Of Request) Call-ID = [EMAIL PROTECTED] (Unique Identifier) CsCseq 102 REGISTER (Command Sequence Number) User-Agent = InInnoMediaTMTA28-2 V2.2.59 SN/001099006807 (Client Information) Contact = sip:[EMAIL PROTECTED]:5060;cos=0;stun=0 (Contact Details) Expires = 3600 (Time After Which Message Content Expires) Proxy-Authorization = Digest ususername711130276219,realm=kukurtururisip:192.168.0.1:5060,nonce=773FCFC316B3EC0E3BFBFABECEAD7957,algorithm=MD5,response=e5ff2cece6dd0509fefe1bcbcf19d8e2 (Client Identification) Max-Forwards = 70 (Limit On Number Of Proxies/Gateways) Content-Length = 0 (Message Body Length In Octets) Kurt __ Do you Yahoo!? New and Improved Yahoo! Mail - 100MB free storage! http://promotions.yahoo.com/new_mail ___ 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] permission problem
Cyprien Simons wrote: Is the only way to use asterisk _not_ as root to change the permission of all the directories where asterisk need to create a file? (/var/run/, /var/log/asterisk/messages) http://voip-info.org/wiki-Asterisk+non-root 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] QoS in asterisk
[EMAIL PROTECTED] wrote: I use IAX trunking and a ping script to check times and fluctuations to my remote offices. Could you share this AGI? - seems like a useful example :) Thanks a lot, 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] How to make * don't strip the leading 0
Roger Schreiter schrieb:
I have currently the same problem with my E1 card and I wonder,
...
SetCallerID(0${CALLERIDNUM})
O.k. this works fine for me too.
I hope, I won't have to take special care, when
calls came from local or from international.
Roger.
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Re: [Asterisk-Users] How to make * don't strip the leading 0
Roger Schreiter [EMAIL PROTECTED] wrote: [...] I have currently the same problem with my E1 card and I wonder, how I can get asterisk to append a leading 0 before forwarding the call, for my IP phones show the correct callee number with leading 0. I ended up just writing a Perl AGI script to canonicalise incoming CLI. -- Hockey has never made much sense to me. In Rugby (my sport of choice, because it's about the only sport where fat, overweight, out of shape guys are actually a sought after commodity), you've got hands, feet, knees, elbows, heads, and teeth. In hockey you've got all that, plus they give you a *STICK*! ___ 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] QoS in asterisk
On Tue, 13 Jul 2004 [EMAIL PROTECTED] wrote: Qualify will only stop the call going through if for example the ping is above 200ms. I find most of my problems come from fluctuating ping times (~100ms) than from a stable high ping. I agree that the overall delay isn't really the problem - jitter and packet loss are what causes the trouble. There really isn't currently anything in Asterisk which measures this - especially not when there is no active call using the path. The IAX2 jitter buffer code does know the amount of jitter - and could probably make this measurement available in a variable or something. And I propose to add similar jitter buffer code for SIP and other RTP-using protocols too. But I'm not really sure how the measurement can then be used effectively for call routing. I'd be interested in your ideas. Note that I observe that in my environment jitter and packet loss come and go over a timescale of seconds - this a result of sharing a narrowish pipe with a bunch of other traffic without any shaping to help the VOIP traffic. For this environment the real fix is to improve the network rather than do anything too complicated with *. (Not to say that *s jitter handling and packet-loss-concealment can't be improved - I've been working on that and I'm still busy). I'm about to ask for some help in gathering jitter stats from a bunch of users - perhaps you'd like to help with that. 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
[Asterisk-Users] ZapBarge and SIP Channels
Hello everybody, Is there any alternative to Asterisk ZapBarge command for SIP and IAX channels? Thanks Lamine
Re: [Asterisk-Users] Stopping reinvite with IAX2?
Thanks for this. I think I have it working as desired. What are the implications of allowing the transfer to occur? I'm not confidetn about allowing my server to lose control of the call. I would be in effect allowing my cell phone to communicate directly with VPC. Can I be certain about call hangup under all circumstances, etc. Thanks, Michael On Sun, 11 Jul 2004 23:30:22 -0500, Brian K. West wrote: per peer bkw - Original Message - From: Michael Graves [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, July 11, 2004 9:25 PM Subject: Re: [Asterisk-Users] Stopping reinvite with IAX2? Is this set on a per peer basis, or in the general section? Michael On Sun, 11 Jul 2004 22:10:26 -0500, Brian K. West wrote: notransfer=yes bkw - Original Message - From: Michael Graves [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, July 11, 2004 9:09 PM Subject: [Asterisk-Users] Stopping reinvite with IAX2? Hi All, I'm using DISA on my * server to avoid overseas toll charges when making calls to Western Europe from my cell phone. I have DISA working with a DID from a VoicePulse Connect account. The outgoing call to Europe is also made via Voicepulse Connect. I see that the IAX media path is bridging the inbound call to the outbound call so that the media stream entirely bypasses my server once the call is established. I would rather not have this happen. With SIP I see that I can disable the reinvite capability. Is there a similar means to defeat the bridging with IAX2? What are my options? Thanks, Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 Plutocrats beware... ** Tag(s) inserted by Bandit Tagger98 - http://www.gbar.dtu.dk/~c918704 ___ 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 -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 It is dangerous to be correct about matters when the established authories are wrong. - Voltaire ** Tag(s) inserted by Bandit Tagger98 - http://www.gbar.dtu.dk/~c918704 ___ 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 -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 The problem with political jokes is that far too often they actually get elected. ** Tag(s) inserted by Bandit Tagger98 - http://www.gbar.dtu.dk/~c918704 ___ 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] Using Cisco AS5350 as pstn GW .. one-way audio problem
What's your relevant dial peer sip.conf config? -g On Fri, 2004-07-09 at 03:49, Mikael Andersson wrote: Glen Hinkle wrote: I assume the pstn is your * system. Can you get audio both ways if you send the traffic back to *? pstn - as5350 - pstn ? -g Iuse the as5350 for termination at my telco, so it's physicly located there. When I call pstn - as5350 - (sip) asterisk, I can hear the audio from the asterisk, but audio from pstn will not get through. I tried: psth -- as5350 -- sipphone. and the same result. I can hear the sipphone but the sipphone cannot hear me. the as5350 is connected to my telco with dual trunked E1's /Micke ___ 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] How to make * don't strip the leading 0
On Mon, 12 Jul 2004 14:57:42 +0200, Kai Militzer [EMAIL PROTECTED] wrote:
Hi folks!
Is it possible to tell asterisk not to strip the leading 0 of *incoming*
MSNs? I use asterisk with i4l and whenever I get a call from an
long-distance party, the leading 0, which should be there according the
german numbering, is not. So if I get a call from a mobile phone
0177-1234567 should be displayed, but 177-1234567 is displayed. I double
checked if I've forgotten to remove an option to strip the first digit
of incoming calls and found nothing.
The wiki and the mailinglist archives can't enlight me either, why
asterisk behaves like this, or how I can turn it off. So if someone
could give me a hint, I would be very delighted!
You could try adding the leading zero.
For example, I have:
[incoming-isdn]
[incoming-isdn]
exten = msn,1,NoOp
exten = msn,2,SetCallerID(0${CALLERIDNAME} 00${CALLERIDNUM})
exten = msn,3,GotoIf,$[${CALLERIDNUM} = 000]?200:4
exten = msn,4,NoOp
exten = msn,5,Goto(local-extensions,7000,1)
exten = msn,200,SetCallerID(Private )
exten = msn,201,Goto(4)
(my number has been replaced with msn)
This adds the leading 0 to calleridname, and 00 to calleridnum (so it
included the '0' needed to dial externally). It has an unfortunate
side effect of setting the caller ID number to '000' if the telco
doesn't send any caller ID (which also happens to be the emergency
number here in Australia), so I have the GotoIf to catch that
condition and replace it with Private.
I don't know how that works for incoming International calls (never
tested), but it works just fine for national calls.
-Shaun
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[Asterisk-Users] Cisco Remote-Party-ID / Bug #2012
Hello Guys, after an update to cvs head (thanks oej!) my CiscoGW can now flag unkown caller's to Number AND Name Unkown. Before i again open a new bug (which isn't a bug :-)), can someone confirm this: - PrivacyManager does not recognize this as an unknown number - it's not possible to set ANY CID with SetCallerID, it allways stays on Unknown (with chan_capi i had to do a SetCallerID() to get PM to recognize it...) - is there a variable with the stat of the privacy indicator in the remote-party-id? - Is there a way to set CALLERIDNUM to an alphanumeric value? I've seen this with asterisk, anonymous and unknown, so it is possible with the Cisco 7960... Thanks and regards, Andreas _ Check out news, entertainment and more @ http://xtra.co.nz/broadband ___ 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] PRI numbering plan
On 12/07/04 11:11, Michael Sandee wrote:
pridialplan=unknown
prilocaldialplan=national
Not only is this that undocumented, but the string prilocaldialplan
doesn't even show up in the latest CVS HEAD source code, so that's not
going to work...
On 12/07/04 13:36, Thomas wrote:
I have an E100P connected to our partner's PBX. They want the
following:
Called number must have numbering plan/type set as:
unknown/unknown and calling number in: ISDN/national.
Our telco requires exactly this same thing - different TON for the calling
and called numbers. You want to apply a patch I wrote that allows you to
configure them separately.
It swaps the single setting pridialplan for two settings that take the
same values as pridialplan: calledpridialplan and callerpridialplan.
I attach the patch (although it is against a pretty old version of
chan_zap.c). I will also clean this up soon and add it to the bug tracker.
Best regards,
Al
--
Alastair Maw
Systems Analyst
Tel: +44 (0) 845 666 7778
http://www.mxtelecom.com
--- chan_zap.c.org 2004-02-20 16:53:31.0 +
+++ chan_zap.c 2004-03-05 12:03:53.0 +
@@ -282,7 +282,8 @@
int minidle;/* Min # of idling calls to keep active */
int nodetype;/* Node type */
int switchtype;/* Type of switch to emulate */
- int dialplan; /* Dialing plan */
+ int callerdialplan; /* Caller dialing plan */
+ int calleddialplan; /* Called dialing plan */
int dchannel; /* What channel the dchannel is on */
int channels; /* Num of chans in span (31 or 24) */
int overlapdial; /* In overlap dialing mode */
@@ -317,7 +318,8 @@
}
static int switchtype = PRI_SWITCH_NI2;
-static int dialplan = PRI_NATIONAL_ISDN + 1;
+static int callerdialplan = PRI_NATIONAL_ISDN + 1;
+static int calleddialplan = PRI_NATIONAL_ISDN + 1;
#endif
@@ -1595,9 +1597,9 @@
}
p-digital = ast_test_flag(ast,AST_FLAG_DIGITAL);
if (pri_call(p-pri-pri, p-call, p-digital ? PRI_TRANS_CAP_DIGITAL : PRI_TRANS_CAP_SPEECH,
- p-prioffset, p-pri-nodetype == PRI_NETWORK ? 0 : 1, 1, l, p-pri-dialplan - 1, n,
+ p-prioffset, p-pri-nodetype == PRI_NETWORK ? 0 : 1, 1, l, p-pri-callerdialplan - 1, n,
l ? (ast-restrictcid ? PRES_PROHIB_USER_NUMBER_PASSED_SCREEN : (p-use_callingpres ? ast-callingpres : PRES_ALLOWED_USER_NUMBER_PASSED_SCREEN)) : PRES_NUMBER_NOT_AVAILABLE,
- c + p-stripmsd, p-pri-dialplan - 1,
+ c + p-stripmsd, p-pri-calleddialplan - 1,
((p-law == ZT_LAW_ALAW) ? PRI_LAYER_1_ALAW : PRI_LAYER_1_ULAW))) {
ast_log(LOG_WARNING, Unable to setup call to %s\n, c + p-stripmsd);
return -1;
@@ -5364,8 +5366,13 @@
free(tmp);
return NULL;
}
- if ((pris[span].dialplan) (pris[span].dialplan != dialplan)) {
- ast_log(LOG_ERROR, Span %d is already a %s dialing plan\n, span + 1, pri_plan2str(pris[span].dialplan));
+ if ((pris[span].calleddialplan) (pris[span].calleddialplan != calleddialplan)) {
+ ast_log(LOG_ERROR, Span %d is already a %s called dialing plan\n, span + 1, pri_plan2str(pris[span].calleddialplan));
+ free(tmp);
+ return NULL;
+ }
+ if ((pris[span].callerdialplan) (pris[span].callerdialplan != callerdialplan)) {
+ ast_log(LOG_ERROR, Span %d is already a %s caller dialing plan\n, span + 1, pri_plan2str(pris[span].callerdialplan));
free(tmp);
return NULL;
}
@@ -5391,7 +5398,8 @@
}
pris[span].nodetype = pritype;
pris[span].switchtype = switchtype;
- pris[span].dialplan = dialplan;
+ pris[span].calleddialplan = calleddialplan;
+ pris[span].callerdialplan = callerdialplan;
pris[span].chanmask[offset] |= MASK_AVAIL;
pris[span].pvt[offset] = tmp;
pris[span].channels = numchans;
@@ -7556,19 +7564,33 @@
}
#endif
#ifdef ZAPATA_PRI
- } else if (!strcasecmp(v-name, pridialplan)) {
+ } else if (!strcasecmp(v-name, calledpridialplan)) {
+ if (!strcasecmp(v-value, national)) {
+calleddialplan = PRI_NATIONAL_ISDN + 1;
+ } else if (!strcasecmp(v-value, unknown)) {
+calleddialplan = PRI_UNKNOWN + 1;
+ } else if (!strcasecmp(v-value, private)) {
+calleddialplan = PRI_PRIVATE + 1;
+ } else if (!strcasecmp(v-value, international)) {
+calleddialplan = PRI_INTERNATIONAL_ISDN + 1;
+ } else if (!strcasecmp(v-value, local)) {
+calleddialplan = PRI_LOCAL_ISDN + 1;
+ } else {
+ast_log(LOG_WARNING, Unknown called PRI dialplan '%s' at line %d.\n, v-value, v-lineno);
+ }
+ } else if (!strcasecmp(v-name, callerpridialplan)) {
if (!strcasecmp(v-value, national)) {
-dialplan = PRI_NATIONAL_ISDN + 1;
+callerdialplan = PRI_NATIONAL_ISDN + 1;
} else if (!strcasecmp(v-value, unknown)) {
-dialplan = PRI_UNKNOWN + 1;
+callerdialplan = PRI_UNKNOWN + 1;
} else if (!strcasecmp(v-value, private)) {
-dialplan = PRI_PRIVATE + 1;
+callerdialplan = PRI_PRIVATE + 1;
} else if (!strcasecmp(v-value, international)) {
-dialplan = PRI_INTERNATIONAL_ISDN + 1;
+
Re: [Asterisk-Users] feature - VM gain adjust?
What about a post processor that performs Compression/Normalization on the recorded voice mail file? On the down side I can see this being a big CPU hog if you are handling a huge amount of calls and trying to normalize a 5 minute long voicemail at the same time. On the upside you don't have to concern yourself determining line loss or similar things. You also wouldn't have to worry about what I call the Seinfeld Syndrome: quit talker / loud talker issues. You would just have two new variables in voicemail.conf - normalization=yes or no and another to set the db value. -Seth On Mon, 2004-07-12 at 08:46, Rich Adamson wrote: Are you suggesting such a thing exists, or that that would be a proposed future application? I propose to think if an AGC / dynamic compressor could be used instead of a config variable. Most sound editors have modules for this. So how would you detect the remote caller is 14.7 db away from * and adjust the 'outbound' voice message to be at some higher audio level? I like the AGC approach, but I'm not sure its realistic in terms of consistently being able to identify the transmission loss from each and every vm call. Since we know what the loss is for each pstn line (to the central office), it would appear that static value would be a good starting point and the user could adjust from there. Much easier (and more likely) to implement. ___ 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 -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ 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] DTMF warning message in log while using SJPhone
I am using the Pocket PC 2003 version of SJPhone and it seems to be working OK. I however do notice hudreds of the following warning message in my asterisk log whenever I use the sjphone: Jul 12 10:37:11 WARNING[-1426744400]: dsp.c:1467 ast_dsp_process: Unable to process inband DTMF on 2 frames My /etc/asterisk/sip.conf: [1234] type=friend host=dynamic dtmfmode=inband username=1234 secret=mypassword nat=yes mailbox=1234 context=intern Something to be worried about? -- Steve Woolley IT Manager ADS Telecom, Inc. 59 Skyline Drive Suite 1250 Lake Mary, Florida 32746 Phone: (407)682-6226 x1110 Fax: (407)682-3455 Cell: (321)229-5311 [EMAIL PROTECTED] www.adstelecom.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] Gogoif with variables acting funny?
Using an example provided by The Hitchhiker's Guide to Asterisk, I
made the following addition to my extensions.conf file:
[inbound-analog]
exten = s,1,Wait(1)
exten = s,2,SetVar(counter=0)
exten = s,3,Answer()
exten = s,4,Wait(1)
exten = s,5,DigitTimeout(15)
exten = s,6,ResponseTimeout(10)
exten = s,7,BackGround(pls-entr-num-uwish2-call)
exten = t,1,SetVar(counter=[${counter}+1])
exten = t,2,Gotoif([${counter}3]?s,7:h,1)
exten = i,1,Playback(invalid)
exten = h,1,hangup()
The hope would be that the pls-entr-num-uwish2-call message would be
offered up to incoming calls 3 times if the caller times out (10
seconds) and then hangup. However the call hangs up 10 seconds after the
first playing of pls-entr-num-uwish2-call.
My asterisk log shows:
-- Executing Wait(Zap/99-1, 1) in new stack
-- Executing SetVar(Zap/99-1, counter=0) in new stack
-- Executing Answer(Zap/99-1, ) in new stack
-- Executing Wait(Zap/99-1, 1) in new stack
-- Executing DigitTimeout(Zap/99-1, 15) in new stack
-- Set Digit Timeout to 15
-- Executing ResponseTimeout(Zap/99-1, 10) in new stack
-- Set Response Timeout to 10
-- Executing BackGround(Zap/99-1, pls-entr-num-uwish2-call) in
new stack
-- Playing 'pls-entr-num-uwish2-call' (language 'en')
-- Timeout on Zap/99-1
== CDR updated on Zap/99-1
-- Executing SetVar(Zap/99-1, counter=[0+1]) in new stack
-- Executing GotoIf(Zap/99-1, [[0+1]3]?s|7:h|1) in new stack
-- Goto (inbound-analog,h,1)
-- Executing Hangup(Zap/99-1, ) in new stack
== Spawn extension (inbound-analog, h, 1) exited non-zero on
'Zap/99-1'
-- Executing Hangup(Zap/99-1, ) in new stack
== Spawn extension (inbound-analog, h, 1) exited non-zero on
'Zap/99-1'
-- Hungup 'Zap/99-1'
It looks to me as if the Gotoif thinks that [0+1] is greater than or
equal to 3 and therefore jumps to hangup.
Am I missing something here?
--
Steve Woolley
IT Manager
ADS Telecom, Inc.
59 Skyline Drive
Suite 1250
Lake Mary, Florida 32746
Phone: (407)682-6226 x1110
Fax: (407)682-3455
Cell: (321)229-5311
[EMAIL PROTECTED]
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Re: [Asterisk-Users] How to make * don't strip the leading 0
The 0 never is there. Check for my post here: http://lists.digium.com/pipermail/asterisk-users/2004-July/053985.html And the solution here: http://lists.digium.com/pipermail/asterisk-users/2004-July/053989.html Kind regards, Martin List-Petersen On Mon, 2004-07-12 at 14:28, Roger Schreiter wrote: Kai Militzer schrieb: ... Is it possible to tell asterisk not to strip the leading 0 of *incoming* MSNs? I use asterisk with i4l and whenever I get a call from an long-distance party, the leading 0, which should be there according the german numbering, is not. So if I get a call from a mobile phone 0177-1234567 should be displayed, but 177-1234567 is displayed. I double Hi, that's rather your ISDN equipment than asterisk, who strips the leading 0. (National numbering scheme) Look at isdnrep! Probably you'll find the same numbers without leading 0 there. I4L forwards those 0-less numbers to asterisk, and asterisk takes them as they are. chap_capi for my AVM Fritz card does display numbers with leading 0. I have currently the same problem with my E1 card and I wonder, how I can get asterisk to append a leading 0 before forwarding the call, for my IP phones show the correct callee number with leading 0. Roger. ___ 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] zaphfc - TE mode - callerid trouble
Thanks for your post, that solved it.
It was just not documented anywhere.
/Martin
On Fri, 2004-07-09 at 15:41, Michael Sandee wrote:
Hi MLP
nationalprefix=0
internationalprefix=00
Regards,
Martin List-Petersen wrote:
I've got a bit trouble with callerid and zaphfc cards.
Basically zaphfc doesn't add the 0 in front of national numbers
(haven't tried a international call yet).
With chan_capi that allways worked fine, however i had to define the
national and international prefixes in capi.conf.
Is there something similar in zapata.conf ?
Here is my zapata.conf:
[channels]
musiconhold=default
;
; ISDN
;
switchtype = euroisdn ; HFC-S TE mode
signalling = bri_cpe_ptmp
prilocaldialplan = national
pridialplan = unknown
echocancel = yes
immediate= yes
group= 1
context = inbound-zap
channel = 1-2
switchtype = euroisdn ; HFC-S NT mode
signalling = bri_net_ptmp
prilocaldialplan = local
overlapdial = no
echocancel = yes
setcallerid = ( ${CALLERIDNUM})
group= 2
immediate= no
context = inbound-internal
channel = 4-5
;
; PSTN
;
signalling = fxs_ks ; X100P
group = 1
echocancel = yes
usecallerid = yes
context = inbound-zap
immediate = no
channel = 7
signalling = fxo_ks ; TDM400
group = 3
context = inbound-internal
immediate = no
channel = 8-11
A d-channel analyzer on the ISDN line gives me a correct setup (beyond
some Eircom specialities, like a truncated called party MSN):
SETUP
Sending complete
Bearer capability
Coding CCITT
Info. transfer capability Speech
Transfer mode/rate Circuit mode, 64 kbps
Channel identification
Interface identificationImplicitly
Interface type Basic interface
Allocation priority Exclusive
Channel B2-channel
Calling party number
Type of number National number
Numbering plan Isdn/telephony (E.164)
Presentation indicator Presentation allowed
Screening indicator Network provided
Number 876218425
Called party number
Type of number Unknown
Numbering plan Isdn/telephony (E.164)
Number 3987
Any suggestions on what could be wrong ?
I have tried different values for prilocaldialplan and pridialplan on
the TE mode HFC-S card, but no joy.
Kind regards,
Martin List-Petersen
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[Asterisk-Users] GnuGK + Asterisk + SIP Provider
Hi guys, I create a topology like fellow: /** / /*** * GK *---* Asterisk *-- Sip Prov * **/ / ***/ || || || H.323 SIP And I wanna configure a setup that the SIP terminal talk with the H323 terminal. For this I would like use the asterisk. My h323.conf file is like: [general] gatekeeper=10.11.2.80 AllowGKRouted=yes [H323Asterisk] type=h323 context=sip_provider prefix=113151,116462 [default] type=h323 context=default My sip.conf file is like: [default] context=default My extension file is like: exten=_1131517400,1,Dial(h323/[EMAIL PROTECTED],10) exten=_1131517401,1,Dial(h323/[EMAIL PROTECTED],10) exten=_2001,1,Dial(h323/[EMAIL PROTECTED],10) exten=_2002,1,Dial(h323/[EMAIL PROTECTED],10) exten=_,1,Dial(sip/[EMAIL PROTECTED],10) exten=_1131517454,1,Dial(sip/[EMAIL PROTECTED],10) exten=_1164626155,1,Dial(sip/[EMAIL PROTECTED],10) exten=_1164626156,1,Dial(sip/[EMAIL PROTECTED],10) [Vocaldata] exten=_1131517454,1,Dial(sip/[EMAIL PROTECTED],30) exten=_1164626155,1,Dial(sip/[EMAIL PROTECTED],30) exten=_1164626156,1,Dial(sip/[EMAIL PROTECTED],30) The result is: My SIP terminal can talk to the H323 terminal, but the H323 terminal cannot call the SIP. Someone confront same problem before?!!? Or someone have an idea about this?!?! Thanks. ps: I am monitoring the network using a sniffer and the gatekeeper don't respond the SIP Provider's invite. But I configure the AcceptUnregistredCalls=1. Giscard ___ Yahoo! Mail agora com 100MB, anti-spam e antivírus grátis! http://br.info.mail.yahoo.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
Re: R: [Asterisk-Users] How to make * don't strip the leading 0
Hi List!
Thanks for the numerous replys. The SetCallerID workaround did it so far
for me. Thank you very much!
Regards
Kai
Am Mo, den 12.07.2004 schrieb Manuel Wenger um 15:24:
Is it possible to tell asterisk not to strip the leading 0
of *incoming* MSNs? I use asterisk with i4l and whenever
I get a call from an long-distance party, the leading 0, which
should be there according the german numbering, is not.
Are you *really* sure that the 0 is transmitted in the CLI, and that it isn't
stripped already by the phone company? I think the easiest thing for you would be to
add the leading 0 before forwarding the call to your SIP client (ie.
SetCallerID(0${CALLERIDNUM}) in your extensions.conf for each extesion where you'd
like to add the 0).
Regards
Manuel
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[Asterisk-Users] Indications missing on Cisco FXO - ATA-186 (SIP)
Calling from a Cisco FXO port to an ATA-186 (SIP 3.1 image) via * (either CVS-HEAD-06/28/04-11:43:41 or CVS-HEAD-07/12/04-15:49:58) I didn't hear any ringing sound get the following on the console: -- Called 5503 -- SIP/5503-f6b5 is ringing WARNING[-1323201616]: channel.c:1375 ast_indicate: Unable to handle indication 3 for 'SIP/10.10.2.250-9903' -- SIP/5503-f6b5 answered SIP/10.10.2.250-9903 Looking at channel.c, I can see that this means that 'condition' is neither of 'AST_CONTROL_PROGRESS' or 'AST_CONTROL_PROCEEDING'. Presumably it's 'AST_CONTROL_RINGING', so why is this not handled? (NB Calls go through fine - all ulaw currently) Thanks a lot, Fran. ___ 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 make * don't strip the leading 0
On Mon, 2004-07-12 at 15:11, Peter Corlett wrote: Roger Schreiter [EMAIL PROTECTED] wrote: [...] I have currently the same problem with my E1 card and I wonder, how I can get asterisk to append a leading 0 before forwarding the call, for my IP phones show the correct callee number with leading 0. I ended up just writing a Perl AGI script to canonicalise incoming CLI. but on your own phone connection you better should get it right. Kind regards, Martin List-Petersen ___ 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] SMDR/CDR - Asterisk integration
iH went to the link to take a look but admin/admin doesn't work - hcir On Jul 9, 2004, at 10:56 AM, San Singhania wrote: Hello everyone, I am developing an online SMDR / call log system for asterisk. This is going to take the form of an executable with embedded sql and webserver, pdf generation, excel generation, graphs. Actually, we have been selling this for a while now with great success and now I am starting work on the integration with Asterisk. Its a windows executbale and the executable is just about 1MB. If someone is interested, let me know. The online demo is at http://demo.callaccounting.ws . The username/password is admin and admin. To print out reports, just leave all the fields for the report selection blank. With regards, San ___ 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] IAXy prov. using DNS
Hi folks, I found that I can config my IAXy to connect to a * server that is has a fixed IP. I'm using dynamic dns solusion, and I want the IAXy to be able to connect to domain.name.server instead of IP. Do you know how to do that? if it is not possible, do you know when will it be? thanks ___ 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
SIP simultaneous registry possible workaround (was Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous registry)
This may sound like a stupid work around, but how about registering
different extensions and putting both of them in the Dial String (so they
would ring at once) and giving both extensions the same caller id?
I do something with my zaptel and x lite phones... I assign them both the
same number and they both come out as the same caller ID. All lines that
you want will ring, plus outgoing caller ID will be what you want it to be.
This gives you also the possibility to have one line which will never be
busy. You pu
I hope this helps,
Youness
ie:
--- in extensions.conf ---:
phone1=SIP/32SIP/33SIP/34
[incoming]
s,1,Dial(${PHONE1})
-- in sip.conf
[phone1]
callerid=Youness Mobile 21
type=friend
secret=secret
[phone2]
callerid=Youness Mobile 21
type=friend
secret=secret
Youness
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[Asterisk-Users] Problem with Capi Channel
Hi all, I have installed a test machine with asterisk in order to try it. I have a problem with capi channel (chan_capi 0.3.4a). When an external call directed to an internal Ip phone is not answered I obtain this warning repeated many times: Jul 12 16:13:43 WARNING[1209214400]: app_dial.c:302 wait_for_answer: Unable to forward frame Jul 12 16:13:43 WARNING[1209214400]: app_dial.c:302 wait_for_answer: Unable to forward frame Jul 12 16:13:43 WARNING[1209214400]: app_dial.c:302 wait_for_answer: Unable to forward frame Jul 12 16:13:43 WARNING[1209214400]: app_dial.c:302 wait_for_answer: Unable to forward frame -- CAPI Hangingup == No one is available to answer at this time -- Executing Busy(CAPI[contr1/492]/1, ) in new stack Sometimes instead the error is the following: Jul 12 16:10:38 NOTICE[1217602880]: chan_capi.c:1172 capi_request: didn't find capi device with outgoing msn = 460. you should check your config! Jul 12 16:10:38 NOTICE[1217602880]: app_dial.c:536 dial_exec: Unable to create channel of type 'CAPI' If the call is answered no problem occurs. Any suggestion? My configuration files are as follow: CAPI.CONF [general] nationalprefix=0 internationalprefix=00 rxgain=1 txgain=1 [interfaces] msn=460 incomingmsn=* controller=1 softdtmf=0 context=sisge echosquelch=1 isdnmode=ptp devices=2 msn=492 incomingmsn=* controller=1 softdtmf=0 context=sisge echosquelch=1 isdnmode=ptp devices=2 SIP.CONF [general] port = 5060 bindaddr = 0.0.0.0 context = sisge tos = lowdelay disallow = all allow = ulaw allow = alaw allow = gsm localnet = 192.168.1.0 localmask = 255.255.255.0 language = it canreinvite= no [492] context=sisge username=492 type=friend secret=492 host=dynamic qualify=yes callerid=492 dtmfmode=rfc2833 EXTENSIONS.CONF [general] static=yes writeprotect=no TRUNK=CAPI [globals] [sisge] exten = 492,1,Dial(SIP/492,60,tr) exten = 492,2,Hangup exten = 492,102,Hangup exten = _.,1,Dial,CAPI/460:bBYEXTENSION exten = _.,2,Hangup ___ 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 = PSTN Pri Causes
hi, we use ser for signalling and asterisk as gateway. is there a possibility to configure the pri-causes for SIP Responses. SER = 404 NOT FOUND = PSTN .. At this moment the Caller gets no connection under this number It would be nice to signalling something like: participant not available at present Asterisk CVS-HEAD-07/07/04-18:53:32 ,same time we checked out libpri. Any ideas? thx, Markus ___ 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] E1 config help and guidance
See inline comments... asterisk wrote: Darren, Many thanks for your help - I've got further, but am still stumped. Have a look at the following table: LED | ISDN| Asterisk --+---+- OOS | Out | Red ACT | Green | Green RED | Out | Red YEL | Out | Out LBK | Out | Out CC| Out | Out DCH | Green | Green The LED indicators are as follows: OOS: Out of Service ACT: Active State RED: Red alarm state detected (Could be Loss of carrier, loss of frame or loss of crc multiframe YEL: Yellow alarm state detected (remote alarm indication from remote end) LBK: Loopback mode CC: Clock controller not equipped DCH: DCH is established. What this table indicates is that if I plug the 2MB pri card from the nortel into the EuroISDN bearer box, the leds light up as shown in the ISDN column. If I plug the nortel card into the asterisk box, then the leds light up as shown in the Asterisk column. I would have expected the leds to light up in the same manner - am I missing something ? Apart from asterisk experience :) Couple of things to look at: 1) Darren's config assumes you have plugged the _whole_ thing together ie PSTN-*-nortel ie 2 e1 connections - your table above implies that you are doing one then the other. This won't work because the config tells * to get the timing source from the PSTN. 2) I've _never_ had any luck using crc4, I always turn it off and only put it back if someone complains. 3) If you have been struggling for a while you may have strayed onto the telco's 'badboy list'. I have one installation where I misconfigured it once and they marked the interface as 'down' at the exchange. Nothing worked untill I rang them, whereupon they marked it as 'up' and things started working (once I'd fixed the config). Tim. Julian -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Storer, Darren Sent: 09 July 2004 21:06 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] E1 config help and guidance Hi Julian, J I want to put asterisk in the middle of our current pbx (Meridian J Option11) Something like this?: - | | PSTN ---span1--| CPE Asterisk NET |--span2--- Nortel | | | | - Assuming that you connect your incoming Telco PRI (PSTN) to span1 and the Nortel PBX to span2 (as depicted above) the lines below should help: Extract from zaptel.conf span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 span=2,0,0,ccs,hdb3,crc4 bchan=32-46,48-62 dchan=47 Extract from zapata.conf pridialplan=local switchtype = euroisdn signalling = pri_cpe group = 1 channel = 1-10 switchtype = euroisdn signalling = pri_cpe group = 2 channel = 32-41 switchtype = euroisdn signalling = pri_net In the config lines above, span1 is set to take timing from the PSTN whilst span2 is configured to give timing to the Nortel. Span1 will behave like a piece of CPE (PBX) and span2 will behave like the NETwork. NB. The channels in group 1 and 2 are depleted as you only have 10 channels enabled on your PRI. After you have implemented the changes above (or any subsequent changes to the low level PRI config) you should, at the very least, remember to restart the Asterisk system or, as Critch advises, power down and up again. HTH Darren -- Comgate TelcoInternetBroadcast -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of asterisk Sent: 09 July 2004 19:00 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] E1 config help and guidance I've googled / voip-info'd / searched until my eyes are blurry, but couldn't see the info I was looking for. I've turned here for help! Asterisk CVS head (9/7/04) Fedora Core 2 (updated to 2.6.6 kernel) DE405P (jumpers set to E1) I want to put asterisk in the middle of our current pbx (Meridian Option11) Currently the meridian has a 2MB pri EuroISDN card linked via a rj-45 into a euroISDN bearer. This bearer only has 10 channels activated (out of the 30). Obviously, this works - handsets make external calls. What I wanted to do was to add * to the mix, in the middle so that it can intercept inbound / outbound calls and do what it needs to do, as well as providing all the extra functionality that this wonderful product provides. In order to achieve this, I assumed that I needed to take rj45 from the bearer box and plug that into span 2, and take a cable from span 1 into the bearer box. My problem (and blurry eyes) come from not understanding the various protocols to assign to each span. I want the meridian to think that it's still plugged into the EuroISDN bearer. So span 2 should be set up as a EuroISDN link ? What should span 1 be set up as ? What channels should be configured ? Any guidance (I'm not looking for the solution (would
[Asterisk-Users] Changed IP and subnet now no SIP Register 403
I built a system and then changed the IP and subnet. Now the phones will not register, getting a 403. Any ideas?
Re: [Asterisk-Users] feature - VM gain adjust?
Hmmm... I don't know if playing with the * code would really be the best here... Although if it was a plug-in app like app_volume or something I guess it couldn't hurt... It really sounds like you have a line issue here. You said that adjusting the gain on your card introduced echo issues. It sounds like you have an impedance mismatch/imbalance. Like your telco is trying to cut corners going from a 4-pair to 2-pair or doing some creative splitting... Do you possibly know where the source of the echo might be coming from? Maybe somewhere under your control? If not it can be a pain getting the telco to acknowledge/fix the problem. Most proprietary PBXs even would have this problem, although they usually don't introduce so much attenuation as your FXO card seems to be doing... I know I know * is way better than a PBX and it should be more flexible. I'm just saying that normally there's no way short of getting the damn telco to fix the problem or getting your own ISDN (T1 if you're in the Telco-Logically backward USA like me) with channel bank... Even then they don't always work... Just my $0.2 ... - Original Message - From: Rich Adamson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, July 12, 2004 5:46 AM Subject: Re: [Asterisk-Users] feature - VM gain adjust? Are you suggesting such a thing exists, or that that would be a proposed future application? I propose to think if an AGC / dynamic compressor could be used instead of a config variable. Most sound editors have modules for this. So how would you detect the remote caller is 14.7 db away from * and adjust the 'outbound' voice message to be at some higher audio level? I like the AGC approach, but I'm not sure its realistic in terms of consistently being able to identify the transmission loss from each and every vm call. Since we know what the loss is for each pstn line (to the central office), it would appear that static value would be a good starting point and the user could adjust from there. Much easier (and more likely) to implement. ___ 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] dsp.c:1467 ast_dsp_process: Unable to process inband DTMF on 2 frames
Hi can anyone help me on this error msg?? dsp.c:1467 ast_dsp_process: Unable to process inband DTMF on 2 frames thnx St ___ 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] feature - VM gain adjust?
At 5:00 PM -0600 on 7/11/04, Rich Adamson wrote: I'm toying with adding a feature request to provide some sort of gain setting for voicemail when accessed from certain interfaces. Maybe something like voicemail=6.0 (db) within a specific channel section of zapata.conf corresponding to a pstn line. Situation: 1. Someone calls into asterisk and leaves a voicemail. The sound is recorded at some volume well below 0 db, and is directly related to the distance asterisk is from the central office (pstn cable loss) plus whatever distance the user placing the call is from his/her central office. 2. I receive a text message that a voicemail was left. 3. I call into asterisk remotely (assume from a cell phone) and retreive the voicemail. My location creates another xx db of loss between myself and asterisk, and voicemail can hardly be heard. Actual Measured Values: 1. Asterisk is 5.6 db from the central office. Called from one pstn line, through the central office, to asterisk and sending a 1004 hz tone at 0db. Recorded the tone into voicemail. (Tone should have been recorded at about 11.2db, two times the cable loss) 2. Called into asterisk again, this time to retreive the voicemail and measured the 1004 hz tone from voicemail. It was -36db actual. This retreival added another 11.2db of loss due to pstn interfaces and plant loss. 3. The calls were through a TDM FXO module with rx and tx gains set to 0. (Changing rx and tx gain to +3 db and repeating the test resulted in a measured -30.5db signal, but these settings create unwanted echo issues. Therefore adjusting channel gain is not an option.) The end result is that retreiving any voicemail message left from a distant location and retreived from a distant location can hardly be heard. By adding the proposed voicemail=6.0 statement to the appropriate channel, any calls connected to voicemail via that channel would effectively increase transmission levels by 6db (or whatever the setting happened to be). In this example case, the setting would increase the vm volume by 12db (or about 24db measured in the above). Anyone have any thoughts on this? Rich Rich - I'll say that this would be very useful. Regardless of where the loss is being inserted, it still exists. I like the idea of associating the voicemail db adjustment on a per-channel basis. I don't want to have to dink around with yelling at the telco to fix something that just works otherwise. Their answer will be Well, turn up the volume on your phone! which is exactly what your proposed patch will do. A simple trial-and-error process should be able to sort out the proper adjustment on any typical system that doesn't have radical db changes across time. I'm heartily in favor of this idea; I'll even throw a donation towards it, if you have a PayPal account. Another cool feature would be app_volume, which would turn up/turn down tx/rx levels dynamically, but that's left for a different day, and after we have an enhanced app_dial that lets single-digit dtmf sequences jump to dialplan routines and then can reconnect bridged calls. See my various rantings about this in months (years!) past. When I get some spare time (ha ha ha) I should really learn how to code this stuff... JT The above feature request has been entered as bug #2023. It also appears that VM has an issue (by itself) with recording/playing volume. Transmitting a 1004hz tone at 0db through a ata186 (set for -1db fxs loss), and then retreiving the same VM results in that tone measured at ~ -10db. Doing the same from a pstn location (via TDM FXO) suggests the same -10db loss (in addition to the pstn loss). Zapata.conf rxgain and txgain set to 0. Using CVS-HEAD-07/12/04, but same result with CVS-HEAD-07/1/04. Entered as bug #2022. Add comments to either if you'd like. ___ 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] feature - VM gain adjust?
On Mon, 2004-07-12 at 09:31, Seth Remington wrote: What about a post processor that performs Compression/Normalization on the recorded voice mail file? On the down side I can see this being a big CPU hog if you are handling a huge amount of calls and trying to normalize a 5 minute long voicemail at the same time. On the upside you don't have to concern yourself determining line loss or similar things. You also wouldn't have to worry about what I call the Seinfeld Syndrome: quit talker / loud talker issues. You would just have two new variables in voicemail.conf - normalization=yes or no and another to set the db value. While I have tried to stay out of the comments here for a while, I would suggest not going post processing. While it might get the problem fixed for now, it isn't a good long term solution. I have experienced similar trouble with recordings from AGI. We have some recordings that where dead on sound wise, and others that ended up being so soft as to be useless. Would it be something people would like to be able to add filters to a line? Consider normalization as a filter. Monitor could then be moved to a filter as well. Echo cancel could be a filter. Set it up so multiple filters could be added and chained together. This could help those with echo chain a couple of filters together and see if that helps. -- Steven Critchfield [EMAIL PROTECTED] ___ 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] zaptel debugging tools
Are there any debugging tools for the digium zaptel cards that would report the activity on the line, such as DTMF and/or connection protocol? I'm looking to debug the connection with a T100P, I don't have $2000 for a T1 test set. Thanks, Glen ___ 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] permission problem
[EMAIL PROTECTED] (Cyprien Simons) writes: Is the only way to use asterisk _not_ as root to change the permission of all the directories where asterisk need to create a file? (/var/run/, /var/log/asterisk/messages) any help will be appreciated, Grab my patches below. It does both chroot and setuid to user asterisk. (You might need to back out one or two of the obvious Openbsd fixes.) I've been running chroot and as user asterisk for a few weeks now on this sip-only server. There are still few loose ends (like music on hold not running correctly, but part of that appears to be an asterisk scheduler problem under OpenBSD that happens even with no chroot etc.) -wolfgang -- Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/ openbsd amd64 http://www.wsrcc.com/wolfgang/ftp/asterisk-openbsd35.patch ___ 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] New Asterisk bounty: SIP simultaneous
in response to Olle's excellent post, ... (B (Bin particular ... (B (BAsterisk is *not* a SIP proxy. It's a SIP registrar and (Blocation server. (BIt's a very clever SIP UA. It wants to be in the middle (Bof the call (Band wants to be in control of each device. This (Bdevice-slave view doesn't match the SIP architecture. (B (Band ... (B (BI've spent a considerable amount of time investigating (Bsupport for (Bmultiple registrations on one Asterisk sip [peer] account (Band after (Blearning about Asterisk's architecture come to the (Bconclusion that (Bit is not an easy or even a desirable feature to (Bimplement. (B (Band ... (B (BIt may be possible, but will probably lead to a lot of (Bchanges to (BAsterisk, both core and applications, that no other (Bchannel will (Bbenefit from. A quick hack to support it may lead to a (Blot of (Bconfusion on how to handle other apps. And it's a lot (Bmore work (Bthan the bounty will cover. I suggest that you use a (Bforking SIP (Bproxy in conjunction with Asterisk to get this (Bfunctionality. (B (BPrecisely! A fairly simple and elegant solution. (B (BFor those rare occasions where one would really need (Bmultiple concurrent SIP registrations I'd say one should (Bconsider running Asterisk in combination with a SIP proxy. (BSince SER is a free download, this wouldn't seem to be (Bsuch a big deal IF IT WASNT for the fact that one will (Bthen need to run two boxes. (B (BIt would make a lot of sense to provide support for an (Beasy-to-configure set up where Asterisk can live together (Bwith another SIP speaking piece of software on the same (Bbox. (B (BSomething along the lines of ... (B (B(ip1:5060)---[*]---[portswapper]---(ip1:5061)---[SER]---(ip2:5060) (B (BSomething like this should allow you to run Asterisk on (Bone address (ie LAN side) and SER on another (ie WAN (Bside), so you get the best of both Asterisk and a SIP (Bproxy all in one box. (B (BThis would also make it possible to run a SIP softphone (Balongside Asterisk on a notebook, so it would solve two (Bbirds with one stone. (B (BI'd like to emphasise however, that most of the problems (Bdescribed in this thread are NOT good reasons for multiple (Bconcurrent SIP registrations. Those problems have other (Bsolutions. Let's take a look at them. (B (B1) Call centre scenario (B (BProblem: multiple agents should receive calls on the same (Bphone number (B (BSolution: assign a number to a call queue and let the call (Bqueue distribute incoming calls to the agents on different (BSIP phones, each of which should have unique logins for (Breasons of accounting and quality assurance. (B (Bmultiple concurrent registrations on the same SIP account (Bin call centres is a BAD IDEA. (B (B2) Overworked admin scenario (B (BProblem: asterisk admin doesn't want to deal with support (Bcalls for adding additional SIP phones (B (BSolution: a simple self provisioning system, either web (Bbased or even IVR based. (B (B3) Dual line desk phone scenario (B (BProblem: dual line desk phone requires multiple (Bregistrations, one per line (B (BSolution: let the phone register on two different SIP (Baccounts, which is how any conventional PBX handles dual (Bline phones: one extension per line. (B (B4) Call group scenario (B (BProblem: multiple phones to ring on the same extension (B (BSolution: use the call group feature or use the dial (Bcommand with multiple SIP peers (B (B (BFor the avoidance of doubt, I am not saying there is no (Bsituation for which multiple concurrent SIP registrations (Bmay be the right solution, but the problems described so (Bfar are *not*. (B (BBut if anybody has a problem that truly warrants parallel (Bforking, then I propose you look into sponsoring somebody (Bto work on the little port swapping trick to run SER (Bconcurrently on your Asterisk box. (B (Brgds (Bbenjk (B (B (B__ (BDo You Yahoo!? (Bhttp://bb.yahoo.co.jp/ (B (B___ (BAsterisk-Users mailing list (B[EMAIL PROTECTED] (Bhttp://lists.digium.com/mailman/listinfo/asterisk-users (BTo UNSUBSCRIBE or update options visit: (B http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Gogoif with variables acting funny?
snip -- Executing SetVar(Zap/99-1, counter=[0+1]) in new stack -- Executing GotoIf(Zap/99-1, [[0+1]3]?s|7:h|1) in new stack -- Goto (inbound-analog,h,1) snip It looks to me as if the Gotoif thinks that [0+1] is greater than or equal to 3 and therefore jumps to hangup. Am I missing something here? I apologize in advance for the stupid question, but is it at all possible that counter is being evaluated in a string context either in the additionor the GoToIf command? (One quick way to check that is to see what happens if you put a second addition in right after the first, and see if you get '2', or '[[0+1]+1]'). Shaun __ Do you Yahoo!? New and Improved Yahoo! Mail - Send 10MB messages! http://promotions.yahoo.com/new_mail ___ 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] PRI numbering plan
On Mon, 2004-07-12 at 15:30, Alastair Maw wrote: On 12/07/04 11:11, Michael Sandee wrote: pridialplan=unknown prilocaldialplan=national Not only is this that undocumented, but the string prilocaldialplan doesn't even show up in the latest CVS HEAD source code, so that's not going to work... prilocaldialplan is not something that is part of asterisk, but introduced in the patches from kapejod's bristuff 0.0.2 (http://www.junghanns.net), which add's BRI zaptel telephony to asterisk. That is the reason why some people have it and some not. Kind regards, Martin List-Petersen ___ 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] X101P FXO with RED alarm
Richard, 1. don't run 0.5 zaptel driver with asterisk-head it will panic the kernel. 2. I am pretty sure that the current BSD zaptel driver only supports the fxs modules and the x100p card. Chris - Original Message - From: Richard Airlie [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, July 11, 2004 11:02 PM Subject: Re: [Asterisk-Users] X101P FXO with RED alarm On Sat, Jul 10, 2004 at 05:55:21PM +0100, Kevin Walsh wrote: Richard Airlie [EMAIL PROTECTED] wrote: First things first. Scrap the ports and build from the latest CVS source. 0.9 is far to old and buggy, and suspect the same of the Zaptel driver you have, although I don't use *BSD myself. I cvsup'd to the latest source yesterday and tried to build zaptel, but it failed right away. (trying to include linux/*.h) I didn't try building asterisk as it seems like the problem is with zaptel -- i.e. I should be able to load the zaptel driver and not see a red alarm, irrespective of my asterisk version, right?. Secondly, the red alarm does tend to mean that the line is not connected, but I got what you're describing when I moved Asterisk to a new machine. Try the X100P card in a different PCI slot. That cleared it for me, for whatever reason. Thanks for that, I gave it a try but unfortunately it's made no difference. I am suspecting the problem is either with the zaptel driver in ports (which is the only version I can get to build) or i've got a hardware issue. For what it's worth I can plug a phone into the back of the FXO and get dial tone, so I guess that proves that the cabling is OK? best, Richard. ___ 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] Sort of OT: Recommended USB handset for use with iaxComm?
One of my coworkers needs to get a softphone set up to my Asterisk system; he's a Linux user, so it looks like about the only IAX2 option is iaxComm. For ease of use (he'll be using this a fair bit), I'm recommending that he get a USB handset; I'm just having trouble finding any US retailers for them. :) Could someone recommend a USB handset that's compatible with iaxComm available for reasonably fast/inexpensive shipment to the US48 area? Thanks! | nate carlson | [EMAIL PROTECTED] | http://www.natecarlson.com | | depriving some poor village of its idiot since 1981| ___ 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] Indications missing on Cisco FXO - ATA-186 (SIP)
Calling from a Cisco FXO port to an ATA-186 (SIP 3.1 image) via * (either CVS-HEAD-06/28/04-11:43:41 or CVS-HEAD-07/12/04-15:49:58) I didn't hear any ringing sound get the following on the console: -- Called 5503 -- SIP/5503-f6b5 is ringing WARNING[-1323201616]: channel.c:1375 ast_indicate: Unable to handle indication 3 for 'SIP/10.10.2.250-9903' -- SIP/5503-f6b5 answered SIP/10.10.2.250-9903 Looking at channel.c, I can see that this means that 'condition' is neither of 'AST_CONTROL_PROGRESS' or 'AST_CONTROL_PROCEEDING'. Presumably it's 'AST_CONTROL_RINGING', so why is this not handled? (NB Calls go through fine - all ulaw currently) Someone else just had that same problem in the last day or two. I don't have their response, but it had something to do with setting the Audiomode to different value to take advantage of a codec or something to that effect. Search the archives... ___ 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] PRI numbering plan
Oh I'm sorry... this setting was probably bri-stuff specific. I didn't
know... I've been using it for a while now and got used to it.
On most ISDN2/BRI lines you need the setting below to actually have a
correctly functioning line (with proper outgoing callerid). It is
probably why it was added for the quad/octobri's.
Considering the situation below it probably should be added in some way
or another.
Alastair Maw wrote:
On 12/07/04 11:11, Michael Sandee wrote:
pridialplan=unknown
prilocaldialplan=national
Not only is this that undocumented, but the string prilocaldialplan
doesn't even show up in the latest CVS HEAD source code, so that's not
going to work...
On 12/07/04 13:36, Thomas wrote:
I have an E100P connected to our partner's PBX. They want the following:
Called number must have numbering plan/type set as:
unknown/unknown and calling number in: ISDN/national.
Our telco requires exactly this same thing - different TON for the
calling and called numbers. You want to apply a patch I wrote that
allows you to configure them separately.
It swaps the single setting pridialplan for two settings that take
the same values as pridialplan: calledpridialplan and
callerpridialplan.
I attach the patch (although it is against a pretty old version of
chan_zap.c). I will also clean this up soon and add it to the bug
tracker.
Best regards,
Al
--- chan_zap.c.org 2004-02-20 16:53:31.0 +
+++ chan_zap.c 2004-03-05 12:03:53.0 +
@@ -282,7 +282,8 @@
int minidle;/* Min # of idling calls to keep
active */
int nodetype; /* Node type */
int switchtype; /* Type of switch to emulate */
- int dialplan; /* Dialing plan */
+ int callerdialplan; /* Caller dialing plan */
+ int calleddialplan; /* Called dialing plan */
int dchannel; /* What channel the dchannel is on */
int channels; /* Num of chans in span (31 or 24) */
int overlapdial;/* In overlap dialing mode */
@@ -317,7 +318,8 @@
}
static int switchtype = PRI_SWITCH_NI2;
-static int dialplan = PRI_NATIONAL_ISDN + 1;
+static int callerdialplan = PRI_NATIONAL_ISDN + 1;
+static int calleddialplan = PRI_NATIONAL_ISDN + 1;
#endif
@@ -1595,9 +1597,9 @@
}
p-digital = ast_test_flag(ast,AST_FLAG_DIGITAL);
if (pri_call(p-pri-pri, p-call, p-digital ? PRI_TRANS_CAP_DIGITAL : PRI_TRANS_CAP_SPEECH,
- p-prioffset, p-pri-nodetype == PRI_NETWORK ? 0 : 1, 1, l, p-pri-dialplan - 1, n,
+ p-prioffset, p-pri-nodetype == PRI_NETWORK ? 0 : 1, 1, l, p-pri-callerdialplan - 1, n,
l ? (ast-restrictcid ? PRES_PROHIB_USER_NUMBER_PASSED_SCREEN : (p-use_callingpres ? ast-callingpres : PRES_ALLOWED_USER_NUMBER_PASSED_SCREEN)) : PRES_NUMBER_NOT_AVAILABLE,
- c + p-stripmsd, p-pri-dialplan - 1,
+ c + p-stripmsd, p-pri-calleddialplan - 1,
((p-law == ZT_LAW_ALAW) ? PRI_LAYER_1_ALAW : PRI_LAYER_1_ULAW))) {
ast_log(LOG_WARNING, Unable to setup call to %s\n, c + p-stripmsd);
return -1;
@@ -5364,8 +5366,13 @@
free(tmp);
return NULL;
}
- if ((pris[span].dialplan) (pris[span].dialplan != dialplan)) {
- ast_log(LOG_ERROR, Span %d is already a %s dialing plan\n, span + 1, pri_plan2str(pris[span].dialplan));
+ if ((pris[span].calleddialplan) (pris[span].calleddialplan != calleddialplan)) {
+ ast_log(LOG_ERROR, Span %d is already a %s called dialing plan\n, span + 1, pri_plan2str(pris[span].calleddialplan));
+ free(tmp);
+ return NULL;
+ }
+ if ((pris[span].callerdialplan) (pris[span].callerdialplan != callerdialplan)) {
+ ast_log(LOG_ERROR, Span %d is already a %s caller dialing plan\n, span + 1, pri_plan2str(pris[span].callerdialplan));
free(tmp);
return NULL;
}
@@ -5391,7 +5398,8 @@
}
pris[span].nodetype = pritype;
pris[span].switchtype = switchtype;
- pris[span].dialplan = dialplan;
+ pris[span].calleddialplan = calleddialplan;
+ pris[span].callerdialplan = callerdialplan;
pris[span].chanmask[offset] |= MASK_AVAIL;
pris[span].pvt[offset] = tmp;
pris[span].channels = numchans;
@@ -7556,19 +7564,33 @@
}
#endif
#ifdef ZAPATA_PRI
- } else if (!strcasecmp(v-name, pridialplan)) {
+ } else if (!strcasecmp(v-name, calledpridialplan)) {
+ if (!strcasecmp(v-value, national)) {
+calleddialplan = PRI_NATIONAL_ISDN + 1;
+ } else if (!strcasecmp(v-value, unknown)) {
+calleddialplan = PRI_UNKNOWN + 1;
+ } else if (!strcasecmp(v-value, private)) {
+calleddialplan = PRI_PRIVATE + 1;
+ } else if (!strcasecmp(v-value, international)) {
+calleddialplan = PRI_INTERNATIONAL_ISDN + 1;
+ } else if (!strcasecmp(v-value, local)) {
+calleddialplan = PRI_LOCAL_ISDN + 1;
+ } else {
+ast_log(LOG_WARNING,
Re: [Asterisk-Users] How to make * don't strip the leading 0
On Mon, 2004-07-12 at 16:09, Martin List-Petersen wrote: On Mon, 2004-07-12 at 15:11, Peter Corlett wrote: Roger Schreiter [EMAIL PROTECTED] wrote: [...] I have currently the same problem with my E1 card and I wonder, how I can get asterisk to append a leading 0 before forwarding the call, for my IP phones show the correct callee number with leading 0. I ended up just writing a Perl AGI script to canonicalise incoming CLI. but on your own phone connection you better should get it right. Hmm .. my email-client cut a line away there (or was it just me ?): It should have said: Quite a lot VoIP providers don't get the CallerID right (especially the national/international issue), but on your own phone connection you better should get it right. Kind regards, Martin List-Petersen ___ 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] Re: Gogoif with variables acting funny?
On Mon, Jul 12, 2004 at 10:51:24AM -0400, Steve Woolley wrote:
exten = t,1,SetVar(counter=[${counter}+1])
exten = t,2,Gotoif([${counter}3]?s,7:h,1)
You need $2
Example:
SetVar(lala=$[1 + 2]);
GotoIf($[${CALLERIDNUM} = 303]?3:2)
http://www.voip-info.org/wiki-Asterisk+Expressions
http://www.voip-info.org/wiki-Asterisk+cmd+GotoIf
--
Stefan Tichy [EMAIL PROTECTED]
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[Asterisk-Users] Cheap ISDN interface + Asterisk what to choose?
Hi All I've been away from Asterisk for some time. I was wdonering what the development status is on this? We've already got a couple of Siemens ISDN phones on an ISDN line, and I was wondering what the development status was for using them with Asterisk? My hope is that it is possible to attach the Asterisk to the existing S0 bus, so that we can use the handsets both via Asterisk and the existing ISDN NTBA. Any help and ideas is appreciated! /Fribse ___ 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] wake-up call script in wiki
On Mon, Jul 12, 2004 at 08:47:12AM +0700, Isianto Istiadi wrote: On Fri, 09 Jul 2004 13:58:30 +1000 Dear Gonzalo Servat, I'm successfully using your wake-up script, but found 1 problem. Other than that it works perfectly good. Thanks man. ^_^ anyway, my problem seems to be the timezone or date problem. I'm using time zone WIT/JAV, But when I run the wake up script, in the * console, it says that it doesn't know my timezone. so I edit the date:manipulate, in the date:maipulate, there's a line JAV + 0700 java, I change it to WIT +0700 java. it works, but the time that I entered using wake-up script, always being added 7 hours later. For example, I put 10:00 it become 05:00 pm. Do you have any idea how to solve this? For the mean time, I edit your configurations a little bit to accomodate the problem, but I can't (haven't understood) how to change the asterisk-voice to accomodate that. For example when I enter 10:00 am, the file for outgoing call has been fixed to the above time, but the asterisk voice stil say 05:00 pm. Sorry for My English, thanks Your English isn't all that bad... Let me make sure I've got it right, though... The script reads back the time correctly when you request a wakeup call, but the call file that's created has a filename that's 7 hours off? I don't know what's wrong, and I may not have too much time to spend looking at it, but I'll see what I can do... Rob -- Rob Fugina, Systems Guy [EMAIL PROTECTED] -- http://www.geekthing.com My firewall filters MS Office attachments. The backup's not over 'til the FAT table sings. ___ 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 T1 question
I currently have a Fractional T1 coming into my site which runs into an adtran device which splits out 10 channels for data in the form of an ethernet interface and 14 analog lines for voice. The ethernet goes directly to a pix firewall. How would I split out the T1 so that it sends a T1 to the asterisk server and still provides the ethernet to the pix. can I use a splitter and use the existing adtran and run a line directly to the T1 card in the asterisk server --or would this mess up the signalling? thanks for your assistance. Curtis -- This mail sent through Horde-Toaster (http://qmailtoaster.clikka.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
RE: [Asterisk-Users] Gogoif with variables acting funny?
Are you using the lastest cvs? If not you have a broken gotoif...
bkw
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Steve Woolley
Sent: Monday, July 12, 2004 9:51 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Gogoif with variables acting funny?
Using an example provided by The Hitchhiker's Guide to Asterisk, I
made the following addition to my extensions.conf file:
[inbound-analog]
exten = s,1,Wait(1)
exten = s,2,SetVar(counter=0)
exten = s,3,Answer()
exten = s,4,Wait(1)
exten = s,5,DigitTimeout(15)
exten = s,6,ResponseTimeout(10)
exten = s,7,BackGround(pls-entr-num-uwish2-call)
exten = t,1,SetVar(counter=[${counter}+1])
exten = t,2,Gotoif([${counter}3]?s,7:h,1)
exten = i,1,Playback(invalid)
exten = h,1,hangup()
The hope would be that the pls-entr-num-uwish2-call message would be
offered up to incoming calls 3 times if the caller times out (10
seconds) and then hangup. However the call hangs up 10 seconds after the
first playing of pls-entr-num-uwish2-call.
My asterisk log shows:
-- Executing Wait(Zap/99-1, 1) in new stack
-- Executing SetVar(Zap/99-1, counter=0) in new stack
-- Executing Answer(Zap/99-1, ) in new stack
-- Executing Wait(Zap/99-1, 1) in new stack
-- Executing DigitTimeout(Zap/99-1, 15) in new stack
-- Set Digit Timeout to 15
-- Executing ResponseTimeout(Zap/99-1, 10) in new stack
-- Set Response Timeout to 10
-- Executing BackGround(Zap/99-1, pls-entr-num-uwish2-call) in
new stack
-- Playing 'pls-entr-num-uwish2-call' (language 'en')
-- Timeout on Zap/99-1
== CDR updated on Zap/99-1
-- Executing SetVar(Zap/99-1, counter=[0+1]) in new stack
-- Executing GotoIf(Zap/99-1, [[0+1]3]?s|7:h|1) in new stack
-- Goto (inbound-analog,h,1)
-- Executing Hangup(Zap/99-1, ) in new stack
== Spawn extension (inbound-analog, h, 1) exited non-zero on
'Zap/99-1'
-- Executing Hangup(Zap/99-1, ) in new stack
== Spawn extension (inbound-analog, h, 1) exited non-zero on
'Zap/99-1'
-- Hungup 'Zap/99-1'
It looks to me as if the Gotoif thinks that [0+1] is greater than or
equal to 3 and therefore jumps to hangup.
Am I missing something here?
--
Steve Woolley
IT Manager
ADS Telecom, Inc.
59 Skyline Drive
Suite 1250
Lake Mary, Florida 32746
Phone: (407)682-6226 x1110
Fax: (407)682-3455
Cell: (321)229-5311
[EMAIL PROTECTED]
www.adstelecom.com
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