Re: [asterisk-users] Asterisk as register server through OpenSIPS
9 jan 2012 kl. 09:02 skrev Ronald Cepres: Hi all, I've been trying to register a SIP user agent to an Asterisk server using OpenSIPS as SIP router. The functionality is working fine. However, Asterisk uses the IP address of the OpenSIPS server as the peer IP address. How can I use the original IP address of the peer without changing the peer's nat=yes? I appreciate any kind of help. Thanks! You propably have NAT=yes in Asterisk. If you turn that off, Asterisk will save the contact provided by the phone which will point directly to the phone, bypassing the OpenSIPS proxy. In order to get Asterisk to use the OpenSIPS proxy outbound as well you need to define it as an outbound proxy. Now, you have to configure NAT support in OpenSIPS since it's the first hop seen from the phone. /O Twitter @oej Web: http://edvina.net -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is it valid to Dial(DAHDI/g0/12345wwwww88888888) on an ISDN trunk?
Its the sending complete IE. I'm using EuroISDN and I also use overlap signalling on this interface. Regards Hans On 2012-01-10 01:12, C F wrote: Exactly which IE message are you trying to push manually? you shouldn't have to do that, it should be done in the configs for you. On Mon, Jan 9, 2012 at 1:45 PM, Johann Steinwendtner steinwendt...@gmx.net wrote: On 2012-01-09 17:46, Alex Villacís Lasso wrote: I am trying to collect information regarding a bug report for Elastix (http://bugs.elastix.org/view.php?id=1146). In this bug, an user has asterisk-1.8.7 and dahdi-2.4.1.2. He is trying to make an outbound call through an ISDN trunk, by placing Dial(DAHDI/g0/12345w) in order to send 12345, then wait a period, then send . I am still waiting for a response on what particular telephony card he uses, and the kind of ISDN setup (T1/E1/BRI) being used, but I want to know: Is this dialstring expected to work with an ISDN trunk? If so, are there any configurations that might cause it to stop working? The user claims that this same dialstring worked with Elastix 1.6 which had dahdi-2.2.0.2 and asterisk-1.4.26.1. Some additional information: the user reports that the dial attempt fails with hangup cause 28. From http://helpdesk.netcentral.co.uk/index.php?_m=knowledgebase_a=viewarticlekbarticleid=293 http://helpdesk.netcentral.co.uk/index.php?_m=knowledgebase_a=viewarticlekbarticleid=293 [^ http://helpdesk.netcentral.co.uk/index.php?_m=knowledgebase_a=viewarticlekbarticleid=293] : Code No. 28 - invalid number format (address incomplete). This cause indicates that the called party cannot be reached because the called party number is not in a valid format or is not complete. Is it actually possible that the code is trying to send a string of 'w's through the ISDN link? Or am I misunderstanding? I'm using 'w' to force sending the 'sending complete' IE in an ISDN setup message. But I don't know the usage of multiple 'w' in the dialstring. regards Hans -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Noise in caller handset when dialing out (with dahdi 2.6.0)
Hi, 1. This patch didn't correct the issue but I'm far from certain that I correctly applied the patch. 2. I took the Hardware Echocan module off my board and it seems to correct the issue. I'll dig deeper to check if I correctly applied the patch and both report here and in DAHLIN-275 ticket. 2012/1/9, Shaun Ruffell sruff...@digium.com: On Mon, Jan 09, 2012 at 01:47:48PM -0600, Shaun Ruffell wrote: On Mon, Jan 09, 2012 at 07:52:02PM +0100, Olivier wrote: Hi, On a brand new system, I met an issue I've never met before. My setup is : debian 6.0.3 asterisk 1.8.8.1 dahdi 2.6.0 libpri 1.4.12 freepbx 2.9.0.4 TE420FB (with hardware EC) This is the very first time I'm using Freepbx and the whole configuration was first generated by a make samples command. When a SIP hardphone dials another SIP hardphone, everything is OK. When the same SIP hardphone dials an external number, there is a quite loud noise that can be heard within the handset from the moment the digit sequence have been sent to the moment any party hangs up. The other party doesn't hear this noise. (It's not easy for me to describe this noise) Hi Olivier, I just received another report of this in the morning and was able to reproduce it. I'll create a JIRA issue here shortly with a patch and reply to this thread with that patch. Sorry about the trouble, Olivier, I've attached a patch to DAHLIN-275 [1] which I believe will resolve this for you. After some more testing it will be rolled into a 2.6.0.1 release. [1] https://issues.asterisk.org/jira/browse/DAHLIN-275 Cheers, Shaun -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connecting to an Old Phone System
On 06/01/12 13:14, Dan Journo wrote: Is there such a thing as an ISDN30e PCI card which can be used with a copy of Asterisk, that can act like a voip gateway between the old phone system, and our asterisk box? Yes Digium sell 2 port PRI cards that support E1. TE200 series. I use them like this to connect to old ISDN PBX. There's no need for an extra box in the middle, just put the ISDN PCI card directly into your Asterisk system. cheers, Paul. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Change port from 5060 on Snom phone
On 06/01/12 16:17, Ishfaq Malik wrote: Hi Does anyone know how to change the target port on a Snom phone. I have tried adding :new port number to the end of the registrar but this doesn't work. It should do. Try putting registrarip:port into Outbound Proxy and leave the Registrar box just set to Registrar. Can you email me off list (since this isn't really Asterisk related and a snom support issue, which I can help with) with some details and ideally a SIP trace? cheers, Paul. Advanced - SIP/RTP - Network identity(port) is something else before anyone suggests it. Thanks in advance Ish -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Change port from 5060 on Snom phone
Can you email me off list (since this isn't really Asterisk related and a snom support issue, which I can help with) with some details and ideally a SIP trace? cheers, Paul. Closing this question with a final message including the [SOLVED] phrase will definitely help the community I think. At least I am interested in knowing the answer or final conclusion, would appreciated that very much. Regards, *José Pablo Méndez* -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Noise in caller handset when dialing out (with dahdi 2.6.0) [SOLVED]
2012/1/10, Olivier oza_4...@yahoo.fr: Hi, 1. This patch didn't correct the issue but I'm far from certain that I correctly applied the patch. I was right to suspect I was wrong : now, after correctly applying the DAHLIN-275 patch, it's working OK (with the EchoCan module plugged-in). Thanks for your lighting fast correction !! 2. I took the Hardware Echocan module off my board and it seems to correct the issue. I'll dig deeper to check if I correctly applied the patch and both report here and in DAHLIN-275 ticket. 2012/1/9, Shaun Ruffell sruff...@digium.com: On Mon, Jan 09, 2012 at 01:47:48PM -0600, Shaun Ruffell wrote: On Mon, Jan 09, 2012 at 07:52:02PM +0100, Olivier wrote: Hi, On a brand new system, I met an issue I've never met before. My setup is : debian 6.0.3 asterisk 1.8.8.1 dahdi 2.6.0 libpri 1.4.12 freepbx 2.9.0.4 TE420FB (with hardware EC) This is the very first time I'm using Freepbx and the whole configuration was first generated by a make samples command. When a SIP hardphone dials another SIP hardphone, everything is OK. When the same SIP hardphone dials an external number, there is a quite loud noise that can be heard within the handset from the moment the digit sequence have been sent to the moment any party hangs up. The other party doesn't hear this noise. (It's not easy for me to describe this noise) Hi Olivier, I just received another report of this in the morning and was able to reproduce it. I'll create a JIRA issue here shortly with a patch and reply to this thread with that patch. Sorry about the trouble, Olivier, I've attached a patch to DAHLIN-275 [1] which I believe will resolve this for you. After some more testing it will be rolled into a 2.6.0.1 release. [1] https://issues.asterisk.org/jira/browse/DAHLIN-275 Cheers, Shaun -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Hang up phone after declined attended transfer
We have a customer who has asked us to change this behavior, but I haven't been able to find a way to do it. Server is Asterisk 1.6 and the phones are SPA 303 and 504. Receptionist gets an outside call, starts an attended transfer The office person being called answers by pressing the speaker button Declines to take the call Receptionist goes back to the original outside call by pressing the line button The office phone goes to hold instead of hanging up If the receptionist hangs presses the hookswitch instead of the line button, then it does hang up the call to the internal office phone, however that phone then goes into reorder tone. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hang up phone after declined attended transfer
You could use a parking lot instead of attended transfer? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Alvarez Sent: Tuesday, January 10, 2012 11:02 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Hang up phone after declined attended transfer We have a customer who has asked us to change this behavior, but I haven't been able to find a way to do it. Server is Asterisk 1.6 and the phones are SPA 303 and 504. Receptionist gets an outside call, starts an attended transfer The office person being called answers by pressing the speaker button Declines to take the call Receptionist goes back to the original outside call by pressing the line button The office phone goes to hold instead of hanging up If the receptionist hangs presses the hookswitch instead of the line button, then it does hang up the call to the internal office phone, however that phone then goes into reorder tone. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hang up phone after declined attended transfer
On Tue, Jan 10, 2012 at 10:24 AM, Danny Nicholas da...@debsinc.com wrote: You could use a parking lot instead of attended transfer? Since that actually is more work than just training all the users to manually hang up, I have a feeling the customer won't be enthusiastic about it. But thanks for the idea, I will ask. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hang up phone after declined attended transfer
On Tue, Jan 10, 2012 at 12:02 PM, Carlos Alvarez car...@televolve.comwrote: We have a customer who has asked us to change this behavior, but I haven't been able to find a way to do it. Server is Asterisk 1.6 and the phones are SPA 303 and 504. Receptionist gets an outside call, starts an attended transfer The office person being called answers by pressing the speaker button Declines to take the call Receptionist goes back to the original outside call by pressing the line button The office phone goes to hold instead of hanging up If the receptionist hangs presses the hookswitch instead of the line button, then it does hang up the call to the internal office phone, however that phone then goes into reorder tone. -- Carlos Alvarez TelEvolve 602-889-3003 When the receptionist presses the hookswitch it should hang up the remote internal phone. Playing the reorder tone is due to a setting on the SPA phone. I had to change this for a client that used the SPA phones and I'm drawing a blank as to which setting. Ryan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hang up phone after declined attended transfer
On Tue, Jan 10, 2012 at 1:57 PM, Ryan Wagoner rswago...@gmail.com wrote: On Tue, Jan 10, 2012 at 12:02 PM, Carlos Alvarez car...@televolve.comwrote: We have a customer who has asked us to change this behavior, but I haven't been able to find a way to do it. Server is Asterisk 1.6 and the phones are SPA 303 and 504. Receptionist gets an outside call, starts an attended transfer The office person being called answers by pressing the speaker button Declines to take the call Receptionist goes back to the original outside call by pressing the line button The office phone goes to hold instead of hanging up If the receptionist hangs presses the hookswitch instead of the line button, then it does hang up the call to the internal office phone, however that phone then goes into reorder tone. -- Carlos Alvarez TelEvolve 602-889-3003 When the receptionist presses the hookswitch it should hang up the remote internal phone. Playing the reorder tone is due to a setting on the SPA phone. I had to change this for a client that used the SPA phones and I'm drawing a blank as to which setting. Ryan I did a quick search and found the setting. Go to the Regional tab and find the Reorder Delay. Change that to 255, which will disable the order tone and cause the phone to hangup. Ryan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hang up phone after declined attended transfer
On Tue, Jan 10, 2012 at 12:01 PM, Ryan Wagoner rswago...@gmail.com wrote: I did a quick search and found the setting. Go to the Regional tab and find the Reorder Delay. Change that to 255, which will disable the order tone and cause the phone to hangup. Thanks, that did it! Now to re-train the receptionist... -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Linux Stun Server
I have been running the windows vovida stun server for some time and has worked without issue, but I really want to run a linux stun server and get away from the windows based one. Anyone have an idea of a good replacment that can be compled on opensuse? Thanks Bryant -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Linux Stun Server
Why don't you just use vovida-linux from sourceforge? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bryant Zimmerman Sent: Tuesday, January 10, 2012 3:12 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Linux Stun Server I have been running the windows vovida stun server for some time and has worked without issue, but I really want to run a linux stun server and get away from the windows based one. Anyone have an idea of a good replacment that can be compled on opensuse? Thanks Bryant -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best non polycom SIP conference room phone
Snom is an OEM of the Konftel. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of brya...@zktech.com Sent: Sunday, January 08, 2012 12:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Best non polycom SIP conference room phone Thank you for your responses. No where did I say I hate polycom phones. I personally do not like their approach to sip as a company. Their audio quality is top notch but for me the rest leaves me wanting. Has anyone used the newer snom conference room phone? Bryant Zimmerman On Jan 8, 2012, at 10:59 AM, C F shma...@gmail.com wrote: I find that the bottom line of all polycom haters is ones inability of comprehending the config files and not in its quality. However check out Panasonic. They make a sip conference phone. On 1/5/12, Carlos Alvarez car...@televolve.com wrote: On Thu, Jan 5, 2012 at 5:10 PM, C F shma...@gmail.com wrote: On Thu, Jan 5, 2012 at 12:19 PM, Bryant Zimmerman brya...@zktech.com wrote: I am looking for a really good SIP conference room phone for use with asterisk. I do not like Polycom at all. You have a really bad taste. There was an interesting flamewar one day in the Asterisk IRC channel over Polycom love/hate. We fall into the hate category here, and hope to never have to deal with them. If there was an SPA-series conference phone, we'd all rejoice. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best non polycom SIP conference room phone
So are the Konftel conference room phones any good? Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 From: Jamie A. Stapleton jstaple...@computer-business.com Sent: Tuesday, January 10, 2012 5:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Best non polycom SIP conference room phone Snom is an OEM of the Konftel. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of brya...@zktech.com Sent: Sunday, January 08, 2012 12:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Best non polycom SIP conference room phone Thank you for your responses. No where did I say I hate polycom phones. I personally do not like their approach to sip as a company. Their audio quality is top notch but for me the rest leaves me wanting. Has anyone used the newer snom conference room phone? Bryant Zimmerman On Jan 8, 2012, at 10:59 AM, C F shma...@gmail.com wrote: I find that the bottom line of all polycom haters is ones inability of comprehending the config files and not in its quality. However check out Panasonic. They make a sip conference phone. On 1/5/12, Carlos Alvarez car...@televolve.com wrote: On Thu, Jan 5, 2012 at 5:10 PM, C F shma...@gmail.com wrote: On Thu, Jan 5, 2012 at 12:19 PM, Bryant Zimmerman brya...@zktech.com wrote: I am looking for a really good SIP conference room phone for use with asterisk. I do not like Polycom at all. You have a really bad taste. There was an interesting flamewar one day in the Asterisk IRC channel over Polycom love/hate. We fall into the hate category here, and hope to never have to deal with them. If there was an SPA-series conference phone, we'd all rejoice. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Odd DTMF problem when receiving calls
Hello, I've been tinkering with Asterisk today for the fun of it, trying to set up my own domain. I've got pretty much everything working, including a DID number that connects to my extension. However, I'm having a problem receiving calls from a particular peer, specifically my office's PBX, which allows dialing directly via SIP. (So, just to clarify, the DID number is not involved in this problematic scenario.) The symptom is that my office PBX demands rfc2833 support, and will immediately disconnect a call if the callee doesn't support it. I've tried resolving this a number of ways and the only way that actually works is to create a peer definition in sip.conf for my office's external SIP IP address. Then calls from my office to my Asterisk service work with no problems. However, the peer section is bare: [corp] type=peer host=corp.example.ip port=5060 context=default That's it. That solves the problem. Copying *every setting* in [global] has no effect. In other words, how I configure the peer doesn't appear to matter very much, only that it has a specific configuration at all. This makes no sense to me. I've attached sanitized SIP debug output, with the following replacements: * corp.example.com and corp.example.ip refer to the SIP PBX at my company. (corp.example.com may refer to either the IP or the domain name, due to an oversight while I was sanitizing. This should not be significant. corp.example.ip always refers to the IP.) * asterisk.tld refers to the domain name for my Asterisk install. * asterisk.ip refers to the IP address my Asterisk install binds to (public and static). * NUMBER is my phone number at the company. * phone.nated.ip refers to the public, NATed IP address of the softphone that is registered with Asterisk for my account. * phone.private.ip refers to the private IP address of the softphone. Note that I have nat=yes and directmedia=no for this device in sip.conf, and calls from my DID number work as well as calls to sip:e...@iptel.org. Looking at the SDP negotiation, and at the telephone-event capability in particular, here is what I see happening: 1. Asterisk advertises telephone-event to my softphone. 2. My softphone advertises telephone-event to Asterisk. 3. Asterisk does not offer telephone-event to the company PBX. 4. The company PBX offers telephone-event to Asterisk. After the ACK represented by step 4, the company PBX immediately issues a BYE to Asterisk. If the company PBX has a peer defined in my Asterisk sip.conf file, Asterisk does offer telephone-event to the company PBX in step 3. I've been up and down this issue for a few hours and I cannot for the life of me determine why simply defining a peer causes Asterisk to offer telephone-event. I have tried specifying dtmfmode=rfc2833 or dtmfmode=auto in [global] and neither change has any effect. As I said above, I've copied every configuration directive in [global] into the peer definition for the company PBX, and calls still work. So I'm at a loss to explain this. The problem does not seem to stem from my configuration, but I'm not entirely sure what else could be the problem... an Asterisk bug perhaps? I don't want to jump to that conclusion since this is my first day tinkering with the software. Perhaps someone more knowledgeable can steer me in the right direction? Thanks, -- Chris Howie http://www.chrishowie.com http://en.wikipedia.org/wiki/User:Crazycomputers If you correspond with me on a regular basis, please read this document: http://www.chrishowie.com/email-preferences/ PGP fingerprint: 2B7A B280 8B12 21CC 260A DF65 6FCE 505A CF83 38F5 IMPORTANT INFORMATION/DISCLAIMER This document should be read only by those persons to whom it is addressed. If you have received this message it was obviously addressed to you and therefore you can read it. Additionally, by sending an email to ANY of my addresses or to ANY mailing lists to which I am subscribed, whether intentionally or accidentally, you are agreeing that I am the intended recipient, and that I may do whatever I wish with the contents of any message received from you, unless a pre-existing agreement prohibits me from so doing. This overrides any disclaimer or statement of confidentiality that may be included on your message. --- SIP read from UDP:corp.example.com:5060 --- INVITE sip:1...@asterisk.tld:5060 SIP/2.0 To: sip:1...@asterisk.tld:5060 From: Chris Howie sip:num...@corp.example.com;tag=2958469 Via: SIP/2.0/UDP corp.example.com:5060;branch=z9hG4bKba6f9d45e934097b67461af5f Call-ID: 1cc376aba0701c847ad96d075c4cc...@corp.example.com CSeq: 1 INVITE Contact: sip:num...@corp.example.com:5060 Max-Forwards: 70 x-inin-crn: 2088742465;loc=%3cRegionDefaultLocation%3e Supported: join, replaces Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER, SUBSCRIBE Accept: application/sdp Accept-Encoding: identity
Re: [asterisk-users] Odd DTMF problem when receiving calls
On Tue, Jan 10, 2012 at 6:00 PM, Christopher David Howie m...@chrishowie.com wrote: I've been up and down this issue for a few hours and I cannot for the life of me determine why simply defining a peer causes Asterisk to offer telephone-event. I have tried specifying dtmfmode=rfc2833 or dtmfmode=auto in [global] and neither change has any effect. As I said above, I've copied every configuration directive in [global] into the peer definition for the company PBX, and calls still work. So I'm at a loss to explain this. The problem does not seem to stem from my configuration, but I'm not entirely sure what else could be the problem... an Asterisk bug perhaps? I don't want to jump to that conclusion since this is my first day tinkering with the software. Perhaps someone more knowledgeable can steer me in the right direction? Wheee. You don't say anything about what 'company PBX' is, so we just have to guess based on your description. Based on your description, your 'company PBX' requires that the endpoints it communicates be registered before-hand. Having a definition for the sip peer in asterisk makes asterisk continually register with the 'company PBX'. So it is not necessarily your case that you think it is, that rfc2833 is required, but rather that for 'company PBX', any sip endpoint must first be registered. Both asterisk and 'company PBX' probably support large numbers of possible DTMF or signaling possibilities, and it's not surprising that you can get away with several possible values. If you do 'sip show peers', with and without the config in your sip.conf (use ; to comment it out and 'sip reload' to commit your changes), you should be able to verify that THIS is the real problem. And no, this is not an asterisk bug. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] No audio available on SIP/172.16.129.13:5060-00000001??
Hello, I am trying to run load on asterisk server(version 1.8.7.1) for the voicemail() application using SIPp tool. I am just running sipp at call rate of 1 cps with the following command: ./sipp -m 9000 -r 1 -rp 1000 -d 45 -max_socket 65536 -sf uac_msg_deposit.xml -i 172.16.129.13 -s 1 172.16.129.14 --trace_err I am trying to deposit 9000 messages in the mailbox of user 1 (given by the -s option) but the following warning is coming on the asterisk server due to which the message does not get deposited into the users mailbox: No audio available on SIP/172.16.129.13:5060-0001?? I have set rtpstart=6000 and rtpend=2 in rtp.conf. Can someone please let me know how to avoid these kind of warnings. Thanks. Shalu Thanks and Regards, Shalu Dhamija Rancore Technologies(P) Ltd. Gurgaon Ph : 0124-4200691 +91-9910995356(M) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No audio available on SIP/172.16.129.13:5060-00000001??
Hi Shalu, How you are invoking call in dialplan. it's completely depends on that. And error show that no voice is there for store in voicemail . On Wed, Jan 11, 2012 at 10:05 AM, shalu dhamija shalu.dham...@rancoretech.com wrote: Hello, I am trying to run load on asterisk server(version 1.8.7.1) for the voicemail() application using SIPp tool. I am just running sipp at call rate of 1 cps with the following command: ./sipp -m 9000 -r 1 -rp 1000 -d 45 -max_socket 65536 -sf uac_msg_deposit.xml -i 172.16.129.13 -s 1 172.16.129.14 --trace_err I am trying to deposit 9000 messages in the mailbox of user 1 (given by the -s option) but the following warning is coming on the asterisk server due to which the message does not get deposited into the users mailbox: No audio available on SIP/172.16.129.13:5060-0001?? I have set rtpstart=6000 and rtpend=2 in rtp.conf. Can someone please let me know how to avoid these kind of warnings. Thanks. Shalu Thanks and Regards, Shalu Dhamija Rancore Technologies(P) Ltd. Gurgaon Ph : 0124-4200691 +91-9910995356(M) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Which SpanDSP version to play with Asterisk 10 and T.38/T.30 gatewaying ?
Hi, Maybe I missed it while checking it, but which spandsp version is recommended to play with Asterisk 10 and T.38/T.30 gatewaying ? I can see both spandsp-0.0.6pre17.tgz and spandsp-0.0.6pre18.tgz here (http://www.soft-switch.org/downloads/spandsp/) but I couldn't find a changelog documenting differences between them. So I prefer to double check ask for recommendations. Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anyone have a reliable T.38 Solution
2012/1/5, Kevin P. Fleming kpflem...@digium.com: On 01/04/2012 12:25 AM, Matt Darnell wrote: Aloha, We are looking to roll a solution that will have the following network layout: ISDN-PRI-- Asterisk-- T.38-- ATA-- Fax Does version 1.8 with the Digium fax driver have this capability? I like 1.8 because it is a long term support version. What ATA's are people using? Any working solutions would be great! What you are looking for is T.38 gateway mode (converting between T.30 over modems on a TDM circuit and T.38 over UDPTL), and the answer is no: Asterisk 1.8 does not have T.38 gateway mode. Asterisk 10 does, and it is supported using SpanDSP and res_fax_spandsp. It is not yet supported by Digium's Fax for Asterisk commercial FAX module. Do you have any idea when Digium's Fax for Asterisk commercial FAX module could roughly become supported ? -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problems faced in load testing of asterisk
Hello, I am trying to run load on asterisk server(version 1.8.7.1) through SIPp tool for the voicemail() application. But I am facing a lot of problems. I tried running 1000 calls from SIPp for 100 subscribers (10 messages for each subscriber). I am using odbc storage for the messages. Following warnings/errors are coming on the asterisk server: Jan 11 11:30:49] WARNING[22924] app.c: Failed to lock path '/var/spool/asterisk/voicemail/default/27/INBOX': File exists [Jan 11 11:30:49] ERROR[22924] app_voicemail.c: Couldn't lock directory /var/spool/asterisk/voicemail/default/27/INBOX. Voicemail will be lost. Sometimes I have seen that .lock file remains in the INBOX folder for a particular subscriber. I wanted to know why this .lock file is not deleted. Is this a bug or I am missing something in the configuration. [ Jan 11 11:30:50] WARNING[22874] app_voicemail.c: Open of sound file '/var/spool/asterisk/voicemail/default/2/INBOX/msg0011.gsm' failed: No such file or directory [Jan 11 11:30:50] WARNING[ 23109] app .c: No audio available on SIP/172.16.129.13:9027-0206?? [ Jan 11 11:30:17] ERROR[22122] res_rtp_asterisk.c: Oh dear... we couldn't allocate a port (x=6460)6460 for RTP instance '0x2aaacd6454c8'. errno 98 I have given the rtp port range as 6000 to 8000 in rtp.conf. Is this not sufficient for running 1000 calls. The SIPp command which I am running is as follows: /sipp -m 10 -r 1 -rp 1000 -d 45 -mp 7000 -p 9000 -bg -max_socket 65536 -sf uac_msg_deposit.xml -i 172.16.129.13 -s 1 172.16.129.14 --trace_err usleep 8 ./sipp -m 10 -r 1 -rp 1000 -d 45 -mp 7004 -p 9001 -bg -max_socket 65536 -sf uac_msg_deposit.xml -i 172.16.129.13 -s 2 172.16.129.14 --trace_err usleep 8 ./sipp -m 10 -r 1 -rp 1000 -d 45 -mp 7008 -p 9002 -bg -max_socket 65536 -sf uac_msg_deposit.xml -i 172.16.129.13 -s 3 172.16.129.14 --trace_err usleep 8 ./sipp -m 10 -r 1 -rp 1000 -d 45 -mp 7012 -p 9003 -bg -max_socket 65536 -sf uac_msg_deposit.xml -i 172.16.129.13 -s 4 172.16.129.14 --trace_err - this way it continues for 100 subscribers. Please suggest. Thanks. Shalu-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
