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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

...ITE 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 0612345678@192.168.0.10. Can the above be done? If so, can anyone give me some hints on how to do this?! Thanks in advance, Ronald Voermans -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060113/fbdfc1da/attachment.htm

...STN 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

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

...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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050823/1d84bff0/attachment.htm

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...| - *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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050819/53aadd31/attachment.htm

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

...ow 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

Hi list, I'm with a little problem on codec negotiation between a cisco827 and asterisk. My sip.conf is like that: [general] port = 5060 bindaddr = 0.0.0.0 context = default amaflags = default allow=g729 allow=gsm allow=alaw allow=ulaw ;disallow=all and cisco like that: dial-peer voice 6 voip destination-pattern 0T session protocol sipv2 session target ipv4:<asterisk-ip>

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

...in 180 Ringing To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <200509271002.21828.hzuehl@athene.dnsalias.org> Content-Type: text/plain; charset="iso-8859-1" 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)...

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:

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