[Asterisk-Users] Callerid on transfer
Hello, Suppose customer A calls attendant. CallerID of A is displayed at the attendant. But, when attendant does a consulted transfer to, let's say, B, the callerID of attendant is displayed at B. When the consulted transfer is succesful, the callerid of attendant is STILL displayed at B. Is it possible to, after a successful transfer change the callerid of the attendant in the callerid of A? Regard, Ronald ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Callerid on transfer
ok, thank you! Regards Ronald -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens C F Verzonden: maandag 13 maart 2006 15:27 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: Re: [Asterisk-Users] Callerid on transfer No, at least not yet. On 3/13/06, Ronald Voermans [EMAIL PROTECTED] wrote: Hello, Suppose customer A calls attendant. CallerID of A is displayed at the attendant. But, when attendant does a consulted transfer to, let's say, B, the callerID of attendant is displayed at B. When the consulted transfer is succesful, the callerid of attendant is STILL displayed at B. Is it possible to, after a successful transfer change the callerid of the attendant in the callerid of A? Regard, Ronald ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Grandstream hold one way audio -URGENT
Hi all, At our customer site i've installed one asterisk server with 20 Grandstream GXP2000's. Firmware 1.0.1.9. When someone dials the customer, the receptionist picks up, and does an attended transfer (the 'grandstream way')to a collegue. Most of the times this goes ok, but sometimes, when the receptionist puts the call on hold, and tries te reconnect to the caller there's only one-way-audio. The receptionist can hear the caller, but the caller cannot hear the receptionist! I've done several ngreps etc. and I can see that traffic is going from asterisk to the receptionist phone, and vice versa. I can predict when this is going to happen: when the receptionist places the call on hold, the caller doesn't hear musiconhold. If the caller does hear musiconhold then everythings goes well. Asterisk states in both occassions that it is starting musiconhold, and again, with ngrep i can see the RTP traffic going from asterisk to the caller and vice-versa. I'm thinking this is a problem with the Grandstream phones, but I'm not sure. I upgraded to of the phones to firmware 1.0.1.12 today, and will contact the customer tomorrow if it had helped. Has anyone ever seen this kind of behavior with Grandstreams/Asterisk? Thx, Ronald ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Codec negotiation
Hi All, I've set up an Asterisk box with a SIP trunk to our PSTN provider. I've configured two incoming phonenumbers. One phonenumber is for voice-calls, the other one for receiving faxes. I want the incoming voice-calls to be coded by the G.729 codec, and the fax-number by G.711. Can I make a codec-negotation based on the called number? If you need more info on this, i can send it to you. Thank you all for your answer(s)! Regards, Ronald Voermans ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Codec negotiation
Florian, What exactly do you mean by seperating traffic in to differt SIP peers? The situation is as follows: I have OpenSer connected to our SIP provider/PSTN Provider (the answer to your question: Enertel). Asterisk registers to OpenSer, which then forwards the call to PSTN. Asterisk registers two numbers at OpenSer; one phonenumber and one faxnumber. I also made two entries in sip.conf. However, the host=... Is the same for both numbers. So incoming calls are always matched to one (1) peer/entry in sip.conf. Hence the problem with negotiating the right codec (g.729 for voice, g.711 for fax). Met vriendelijke groet, --- R.L.L.M. Voermans Access Hosting E-mail: [EMAIL PROTECTED] Tel.: +31 (0)161 - 88.88.88 Fax: +31 (0)161 - 88.88.99 Global-e Raadhuisstraat 32 5126 CJ GILZE http://www.global-e.nl --- -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens Florian Overkamp Verzonden: donderdag 9 februari 2006 18:27 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: Re: [Asterisk-Users] Codec negotiation Hi, Ronald Voermans wrote: I've set up an Asterisk box with a SIP trunk to our PSTN provider. I've configured two incoming phonenumbers. One phonenumber is for voice-calls, the other one for receiving faxes. I want the incoming voice-calls to be coded by the G.729 codec, and the fax-number by G.711. Can I make a codec-negotation based on the called number? Nope, but maybe you could separate the traffic in to different SIP peers. If you need more info on this, i can send it to you. If you want we could figure something out. Just curious: Which PSTN provider are you using ? Florian ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Codec negotiation
Yes, But without going deeper into OpenSer (since this IS a Asterisk list): With OpenSer I'm using RTPPRoxy. I don't think i can manage rtpproxy to bind to multiple addresses. I'll look for that anyway. Thanks, Regards, Ronald. -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens Florian Overkamp Verzonden: donderdag 9 februari 2006 23:38 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: Re: [Asterisk-Users] Codec negotiation Hi Ronald, Ronald Voermans wrote: What exactly do you mean by seperating traffic in to differt SIP peers? The situation is as follows: I have OpenSer connected to our SIP provider/PSTN Provider (the answer to your question: Enertel). Ah 'kay. Asterisk registers to OpenSer, which then forwards the call to PSTN. Asterisk registers two numbers at OpenSer; one phonenumber and one faxnumber. I also made two entries in sip.conf. However, the host=... Is the same for both numbers. So incoming calls are always matched to one (1) peer/entry in sip.conf. Hence the problem with negotiating the right codec (g.729 for voice, g.711 for fax). Hrm, yes for inbound the problem is with the host=.. matching. Maybe Olle has a good suggestion on this :-P. However, if you control the OpenSer yourself you could easily bind another IP, or perhaps use OpenSer rules to do the trick ? Asterisk SIP stack doesn't seem suited for this type of traffic separation I guess... Florian ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Use Grandstream ATA as trunk
Hi All, I have a GSM box, which needs to connect to a analogue phone line. I've plugged the GSM box to a Grandstream ATA (386). This ATA has extension number 600. Now what I want to accomplish is the following: - If a mobile-number is chosen by a user, asterisk needs to call the ATA (600), wait for a few seconds, and then send the mobile-phonenumber. Or, if it's possible, define the ATA as a trunk, and then send an INVITE message to the ATA. For example: say the ATA has IP address 192.168.0.10, and I want to make a call to 0612345678; Asterisk sends out an INVITE like INVITE [EMAIL PROTECTED]. Can the above be done? If so, can anyone give me some hints on how to do this?! Thanks in advance, Ronald Voermans ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Empty ACK
Hello, I have asterisk connected to SER/RTPProxy which is again connected to a IP-PSTN gateway. When calling with a UA, registered at * to a SIP phone connected to the IP-PSTN gateway, I get 'empty ACKs': U 192.168.0.173:5060 - 10.254.254.1:5060 ACK SIP/2.0. Via: SIP/2.0/UDP 192.168.0.173:5060;branch=z9hG4bK5cb7d048. Route: sip:[EMAIL PROTECTED]:5060,sip:212.241.48.70:5060. From: 0161801019 sip:[EMAIL PROTECTED];tag=as628d39c1. To: sip:[EMAIL PROTECTED];tag=00-04094-52dc5953-7c1293c27. Contact: sip:[EMAIL PROTECTED]. Call-ID: [EMAIL PROTECTED] CSeq: 103 ACK. User-Agent: Asterisk PBX. Content-Length: 0. As you can see, there is no URI after the ACK statement, and SER doesn't know what to do with it. Is this a bug in *, or is this normal? Regards, Ronald ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Empty ACK
Hello, I have asterisk connected to SER/RTPProxy which is again connected to a IP-PSTN gateway. When calling with a UA, registered at * to a SIP phone connected to the IP-PSTN gateway, I get 'empty ACKs': U 192.168.0.173:5060 - 10.254.254.1:5060 ACK SIP/2.0. Via: SIP/2.0/UDP 192.168.0.173:5060;branch=z9hG4bK5cb7d048. Route: sip:[EMAIL PROTECTED]:5060,sip:212.241.48.70:5060. From: 0161801019 sip:[EMAIL PROTECTED];tag=as628d39c1. To: sip:[EMAIL PROTECTED];tag=00-04094-52dc5953-7c1293c27. Contact: sip:[EMAIL PROTECTED]. Call-ID: [EMAIL PROTECTED] CSeq: 103 ACK. User-Agent: Asterisk PBX. Content-Length: 0. As you can see, there is no URI after the ACK statement, and SER doesn't know what to do with it. Is this a bug in *, or is this normal? Regards, Ronald ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Early Media in 100 Ringing
Hello, I have a problem with the following: When I dial a PSTN number from a UAC, the call is made through a SIP Trunk (which has a connection to the PSTN) in Asterisk. The PSTN Gateway returns a 100 Ringing WITH SDP, but Asterisk forwards the 100 Ringing WITHOUT SDP: As you can see below, the SIP message from 10.254.254.1 (the PSTN Gateway) has SDP, while * (with 192.168.0.173) removes the SDP content. How can this be solved? U 10.254.254.1:5060 - 192.168.0.173:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 192.168.0.173:5060;rport=5060;branch=z9hG4bK454e2d35. Record-Route: sip:[EMAIL PROTECTED]:5060. Record-Route: sip:[EMAIL PROTECTED]:5060;lr;nat=yes. From: 0161801019 sip:[EMAIL PROTECTED];tag=as02de1b95. To: sip:[EMAIL PROTECTED];tag=00-04094-52dbe3bc-6cf68a723. Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE. Contact: sip:212.241.48.70:5060. server: Cirpack/v4.38f (gw_sip). Allow: UPDATE, REFER. Content-Type: application/sdp. Content-Length: 253. . v=0. o=cp10 112775383044 112775383045 IN IP4 10.166.38.109. s=SIP Call. c=IN IP4 10.254.254.1. t=0 0. m=audio 35058 RTP/AVP 18 101. b=AS:64. a=rtpmap:18 G729/8000/1. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000/1. a=fmtp:101 0-15. a=ptime:20. # U 192.168.0.173:5060 - 192.168.1.103:5062 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 192.168.1.103:5062;branch=z9hG4bKff31d98edbf2b265. From: 411 sip:[EMAIL PROTECTED];tag=f93ee2f65c6906cb. To: sip:[EMAIL PROTECTED];tag=as675f246d. Call-ID: [EMAIL PROTECTED] CSeq: 60590 INVITE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER. Contact: sip:[EMAIL PROTECTED]. Content-Length: 0. . ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Early Media in 180 Ringing
If guess I figured it out already. I made some changes in chan_sip.c (when ringing was received, it didn't check for SDP), and recompiled. It's working now! Ronald - -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens Hauke Zuehl Verzonden: dinsdag 27 september 2005 10:02 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: Re: [Asterisk-Users] Early Media in 180 Ringing Hi :) Am Montag, 26. September 2005 19:48 schrieb Ronald Voermans: Hello, As you can see below, the SIP message from 10.254.254.1 (the PSTN Gateway) has SDP, while * (with 192.168.0.173) removes the SDP content. How can this be solved? Well, I am not that expert but AFAIK your PSTN gateway should send a 183 (Session progress) than a simple 180. Do you use Dial(SIP/blah|30|m(moh_class)) to start early media? Regards, Hauke ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Early Media in 180 Ringing
Hello, I have a problem with the following: When I dial a PSTN number from a UAC, the call is made through a SIP Trunk (which has a connection to the PSTN) in Asterisk. The PSTN Gateway returns a 180 Ringing WITH SDP, but Asterisk forwards the 100 Ringing WITHOUT SDP: As you can see below, the SIP message from 10.254.254.1 (the PSTN Gateway) has SDP, while * (with 192.168.0.173) removes the SDP content. How can this be solved? U 10.254.254.1:5060 - 192.168.0.173:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 192.168.0.173:5060;rport=5060;branch=z9hG4bK454e2d35. Record-Route: sip:[EMAIL PROTECTED]:5060. Record-Route: sip:[EMAIL PROTECTED]:5060;lr;nat=yes. From: 0161801019 sip:[EMAIL PROTECTED];tag=as02de1b95. To: sip:[EMAIL PROTECTED];tag=00-04094-52dbe3bc-6cf68a723. Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE. Contact: sip:212.241.48.70:5060. server: Cirpack/v4.38f (gw_sip). Allow: UPDATE, REFER. Content-Type: application/sdp. Content-Length: 253. . v=0. o=cp10 112775383044 112775383045 IN IP4 10.166.38.109. s=SIP Call. c=IN IP4 10.254.254.1. t=0 0. m=audio 35058 RTP/AVP 18 101. b=AS:64. a=rtpmap:18 G729/8000/1. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000/1. a=fmtp:101 0-15. a=ptime:20. # U 192.168.0.173:5060 - 192.168.1.103:5062 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 192.168.1.103:5062;branch=z9hG4bKff31d98edbf2b265. From: 411 sip:[EMAIL PROTECTED];tag=f93ee2f65c6906cb. To: sip:[EMAIL PROTECTED];tag=as675f246d. Call-ID: [EMAIL PROTECTED] CSeq: 60590 INVITE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER. Contact: sip:[EMAIL PROTECTED]. Content-Length: 0. . ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: [Serusers] SER IP PBX for multiple clients
Waldo, How do you let your customers manage 'their' PBX. I too have a setup like you. However, I installed a * server for each customer, via vserver. I'd like to now what kind of software/webbased package you use for this. I also have SER installed as a front-end server for the * servers. But, as I'm still not very into SER, don't know exactly how this fits in. Should I use SER only as proxy, or also as a registrar server (with the same problems as you describe)? Hope someone at this list is able to help us! Regards, Ronald -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens Waldo Rubinstein Verzonden: woensdag 24 augustus 2005 17:28 Aan: Iqbal CC: SER User Mailing List Mailing List; Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: [Asterisk-Users] Re: [Serusers] SER IP PBX for multiple clients lqbal, I do plan on having alot of users. Two markets I'm trying to get some volume users from are: residential consumers and business users. Residential consumers should get basic line services such as their own DID, voicemail, caller-id, call-waiting, three-way calling, and basically, all the standard features you get from companies like Vonage, etc. This particular market base will have a higher volume than business users. Business users will get everything residential consumers get, plus additional features. Features such as, automated attendant, extension- to-extension calling, company directories, etc. I guess I would need to have SER and Asterisk work in tandem. Now, what should be the correct approach in assigning responsibilities to both SER and Asterisk respectively? Should SER be used strictly as proxy to Asterisk, may be also registrar, and NAT helper, and then have Asterisk handle all the calling plans, features, enhanced services and SER will simply forward everything to Asterisk? Can you or someone advise as to what would be the more robust/scaleable architecture to deploy this? Needless to say, it is imperative that I get proper CDR from either one or both systems in order for me to properly bill our users. I don't know which of the two platforms has a more robust/customizable call logging facility. I took the liberty of cross-posting to the Asterisk list in order to get some of their feedback as well. Thanks, Waldo On Aug 23, 2005, at 6:49 AM, Iqbal wrote: Um..no actually I am saying you could combine both, but that will only help if you have alot of users. I guess you could direct calls to a particular sip client, ut normally when ser and asterisk work in tandem, all calls from SER hit one section of sip.conf, and hence can only be pointed to one context, you can get around this by including contexts from this default one, which is what I do, based upon a mysql lookup, but then you will have problems in call pickup, because all pickup is not context based, again there is a solution to this, if you look at bristuff patch for asterisk. If you dont have many users stick with ust asterisk, if you want to scale you may need to kludge something with ser and asterisk, and this might be easy or hard depending on exacly what you require, and call scenarios. Iqbal Waldo Rubinstein wrote: The way I manage this in Asterisk is every SIP UA has a unique login but in different contexts. I suppose that if SER directs a call to Asterisk to the specific SIP client, Asterisk will recognize it belongs to a different context. The question is, I don't know if SER knows about multiple contexts under the premise of the Asterisk world. Also, I get the feeling you are pretty much telling me to stick to Asterisk :) Is that so? Thanks, Waldo On Aug 22, 2005, at 3:26 PM, Iqbal wrote: Hi If you are already using multiple contexts within asterisk, then your already half way there, the problem is if you stick in SER, bcause then your phones are not registered in asterisk, hence all fall into the same context in sip.conf, which means they all will hit one context in extensions.conf, hence you should look into that. I am not sure if you can do the 101/102 extension thing in asterisk, since aliases will be bound to a contact, whereas in asterisk the context is also part of the dialing plan. DID can be done, as can forking and directing to voicemail on no answer. Iqbal Waldo Rubinstein wrote: Hello, I'm still trying to learn more about SER. I've been using Asterisk to manage virtual PBX services for different companies by using multiple contexts within Asterisk. However, since I only use Asterisk with SIP UAs and to communicate with ITSPs, I don't have the need to have all the fancy features Asterisk offers, plus I have the additional advantage of having the built-in NAT support in SER. The question I have is if someone can point me to the right place where I can see some sample configs that do more or less the things I need or
[Asterisk-Users] Music On Hold + canreinvite=yes
For canreinvite=yes to work, I think I need to remove the t argument in the Dial(SIP/ext|60|t) application. Otherwise, Asterisk will allways stay in the middle. I don't want that, so I removed the 't' argument. That works. Now, when two UA are calling, Asterisk gets out of the RTP stream. However, when removing the 't' argument, the Music On Hold doesn't work anymore between these two UA. If I put one UA on hold, Asterisk states that it is starting Music On Hold, but the holding party doesn't hear the audio stream. Is this resolvable? Thanks, Ronald Voermans ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Nat + Asterisk + Ser (Far end Nat Traversal)
Chris, Thank you for your answer. By the way, my * server won't be a PSTN gateway. The SER is connected to another SIP gateway provided by our Telco. Would you be so kind to give me some more details on this: - Say I have * server A with extensions 100, 101, 102, and * server B also with extensions starting from 100, (Server A is for another companyas Server B) How can I make these phones register themselves with SER. Or do I need a unique username for each extension (say username: companyA100, ... and username companyB100) How will I forward all the calls to a * server. So how do I forward the calls from companyA100 to * server A etc... Do the * servers need to be registered at the SER server? Can you please provide me with some example configs (both on the SER as on the * side? I cannot find to many examples on this one (besides the 'Asterisk at large'-WIKI)... Thank you very much, Ronald Voermans Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens Chris HARIGAVerzonden: maandag 22 augustus 2005 15:10Aan: 'Asterisk Users Mailing List - Non-Commercial Discussion'Onderwerp: RE: [Asterisk-Users] Nat + Asterisk + Ser (Far end Nat Traversal) Hi, all your phones need to be registerwith SER. The asterisk will be just PSTN gateway, voicemail server or something else (I prefer to forward all the calls from ser to asterisk because it's easy to manage the dialplan). I have the same configuration,I balance the traffic with SER and I use realtime with asterisk servers. Best regards, Chris HARIGA ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Music On Hold + canreinvite=yes
I found the problem. The ztdummy wasn't loaded. So it had no timer there. When the RTP stream was going through asterisk, I think * used the stream for timing. Ronald -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens Matthew Boehm Verzonden: dinsdag 23 augustus 2005 18:02 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: Re: [Asterisk-Users] Music On Hold + canreinvite=yes Kevin P. Fleming wrote: Matthew Boehm wrote: Umm.. DUH! If you remove the RTP stream from asterisk, asterisk can't send audio (the rtp stream) to the phones. Umm. DUH! Yes it can. When a SIP endpoint is placed on hold, Asterisk will re-INVITE the audio stream back to itself for precisely that reason. Hmm..I stand corrected. And now that I think about it, it seems I jumped the gun without thinking. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
FW: [Asterisk-Users] Nat + Asterisk + Ser (Far end Nat Traversal)
Hello, I have several * serversbehind a SER server (in a local ip range).The SERserveris also publicy reachable. On the other site, I have SIP clients that are behind another NAT or in the same NAT range as the * server. Can someone give me some directions/hints etc. on how to make this work. I think I should be using MediaProxy with SER. But do the SIP clients need to register at the SER server? If not, how will the reach the * server, since they're only reachable VIA the SER router. Here's is scheme: -IP Phone A (Behind NAT router) (ext 100, Asterisk A) - *A-|priv. addr publ. addr| - |--- INTERNET | - SER ---| - |---| - *B-|IP Phone B (Behind NAT router) (ext. 100, Asterisk B) - (Asterisk servers) (10.254.254.x) Phone A can belong to Asterisk A, and B to Asterisk B. Hope this give you enough information. Regards, Ronald Voermans ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Nat + Asterisk + Ser (Far end Nat Traversal)
Hello, I have several * serversbehind a SER server (in a local ip range).The SERserveris also publicy reachable. On the other site, I have SIP clients that are behind another NAT or in the same NAT range as the * server. Can someone give me some directions/hints etc. on how to make this work. I think I should be using MediaProxy with SER. But do the SIP clients need to register at the SER server? If not, how will the reach the * server, since they're only reachable VIA the SER router. Here's is scheme: -IP Phone A (Behind NAT router) (ext 100, Asterisk A) - *A-|priv. addr publ. addr| - |--- INTERNET | - SER ---| - |---| - *B-|IP Phone B (Behind NAT router) (ext. 100, Asterisk B) - (Asterisk servers) (10.254.254.x) Phone A can belong to Asterisk A, and B to Asterisk B. Hope this give you enough information. Regards, Ronald Voermans ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk (multiple) + Ser
I have several Asterisk servers installed and one SER server which will act as a gateway to PSTN, en redirect server. I was thinking to implement it the following way: - Register all the * servers at SER (is this neccessary?) - this works via register=asterisk:[EMAIL PROTECTED] in sip.conf - Setup aliases in SER for the telephonenumbers to the appropiate * server: serctl alias add [EMAIL PROTECTED] [EMAIL PROTECTED] e-mailaddress This way, when one SIP phone behind a * server calls for example 016234567, the * server forwards the request to SER, SER looks up the alias en then forwards it to the destined * server. If a number cannot be handled, SER will forward it to the PSTN gateway. Now my problems: I'm a totaly newby on SER. I managed to get the * server register themselves with SER, and setup Aliases. However I cannot get ser.conf configured so that it does what i've explained before. Is anybody willing to help me out, if possible with a sample ser.conf? TIA, Ronald Voermans ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Multiple Asterisk Installations + SER
If I install 1 * server, with multiple companies/dialplans, how do I make 1 company dial the other company with a full telephonenumber (i.e. 10 digits)? -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens Tzafrir Cohen Verzonden: maandag 15 augustus 2005 8:28 Aan: asterisk-users@lists.digium.com Onderwerp: Re: [Asterisk-Users] Multiple Asterisk Installations + SER On Mon, Aug 15, 2005 at 07:54:50AM +0200, Ronald Voermans wrote: Okay, First of all, thank you for your input. I didn't know that I could use 1 * for multiple companies (wish I knew it earlier, because installing vserver and installing * on a vserver took me a lot of time :) ). Nevertheless, I think I still will need the SER. If my 'shared *' server is getting overloaded, I want to be able to quickly add a new * server. For the IP Voice Interconnect to work properly, I think I need one 'gateway' on our side, which will be SER. Is this correct? Those asterisk instances still share quite a few resources: the network bandwidth and probably the CPU time. With some scriptology, it would probably be rather simple to add another company to your Asterisk configuration. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Multiple Asterisk Installations + SER
I'm not sure I understand what you mean... I want to have internal extensions (100, 101, 102, etc.) and some full phone-numbers (10 digits). How do I implement this in *? Ronald -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens Florian Overkamp Verzonden: maandag 15 augustus 2005 11:58 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: Re: [Asterisk-Users] Multiple Asterisk Installations + SER Hi, Ronald Voermans wrote: If I install 1 * server, with multiple companies/dialplans, how do I make 1 company dial the other company with a full telephonenumber (i.e. 10 digits)? This is very much dependant on how your dialplan works. We use normalisation for each account so the system doesn't have to worry about many different dialling formats (i.e. with or without areacode, and such). You can use a similar strategy for all your internal numbers as well. Florian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Multiple Asterisk Installations + SER
I'm trying to implement a shared asterisk server for multiple (different) companies. Here's what I've done so far: - I've installed multiple asterisk instances on one server (via vserver). Each * is for one customer, and has it's own extensions (like 100, 101, 102, etc.) Note that the same extension can exist on other * instances - The SIP Clients register themselves with * - As a front-server I've installed a SER server. - Via our Telco we have a IP Voice Interconnect, which is our gateway to the PSTN. Can you please advise me what's best to do next: - Let the * servers register themselves to SER - Make aliases for the (public) phonenumbers to the * 'user' - If an PSTN phonecall comes in from our gateway, it comes at SER; SER looks in the aliases to which * it belongs, and forward the call the * which then forwards it to the SIP client (IP Phone/Application...) - Calls made by a SIP client are being handled by *, which forwards the call to SER. SER looks if it's a local call (= handled by A * server) or a PSTN call, and based on that forwards the call to a * server, or to the PSTN gateway. Is this an efficient setup? Our customers our connecting to the * via WAN. Is it smarter to let the SIP Clients register with SER (can they still have the same extensions)? If anyone has some ideas about this, or other suggestions: they're more than welcome! Regards, Ronald Voermans ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Multiple Asterisk Installations + SER
Okay, First of all, thank you for your input. I didn't know that I could use 1 * for multiple companies (wish I knew it earlier, because installing vserver and installing * on a vserver took me a lot of time :) ). Nevertheless, I think I still will need the SER. If my 'shared *' server is getting overloaded, I want to be able to quickly add a new * server. For the IP Voice Interconnect to work properly, I think I need one 'gateway' on our side, which will be SER. Is this correct? Met vriendelijke groet, - R.L.L.M. Voermans Manager Network Connectivity Intern Global-e Raadhuisstraat 32 5126 CJ Gilze (NL) T: +31-(0)161-88 F: +31-(0)161-99 E: [EMAIL PROTECTED] W: http://www.global-e.nl - -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens Matthew Boehm Verzonden: zondag 14 augustus 2005 18:41 Aan: Asterisk Users Onderwerp: Re: [Asterisk-Users] Multiple Asterisk Installations + SER - I've installed multiple asterisk instances on one server (via vserver). Each * is for one customer, and has it's own extensions (like 100, 101, 102, etc.) Note that the same extension can exist on other * instances This is completely UNNECESSARY if you simply use contexts. We have 1 asterisk server running 6 different companies and a good majority of their extensions overlap. This is very easy to configure. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
