SIP based Zimlet?

[skwdenyer]

Hi there

Just seen your post.

Our sipX integration got stalled with other, more critical fish to fry. As, frankly, did Zimbra development - it just wasn't flexible enough (at the time) to do what we needed it to do, and our users just couldn't get on with the web interface. We're now revisiting it, but our focus for the moment is to get a generic version of the "toaster" sorted to integrate Zimbra with another (industry-specific) open source project.

sipX is still on the back burner, and we'll certainly come back to this. The other app also supports click-to-call, so we'll have to start with making a decision as to what we do - Zimbra or not. If we use Zimbra for the calling functionality, the "toaster" will be needed to translate clicks elsewhere into Zimbra calls.

Ho hum, so many decisions, so many variables, so little time :-)

[ByteofKnowledge]

Hi, n00b at both Ubuntu and Zimbra here. Running 4.5.3 on 6.10 Edgy.

We have a Cisco CME system (CallManager Express) in our office, and would be very interested in running a "click 2 call" zimlet to allow our workers to call from their phones at the click of a mouse. Our Zimbra environment is not yet "production", we're in the process of testing it and installing any features we want to add before it goes global.

If anyone would like to help us out, especially bearing in mind my n00b status, please let me know. Maybe it will help out the writers of this zimlet too.

[grazman]

I haven't tried yet. While we can do "click to call" stuff, I haven't found enough information to give me the feeling the SIP Zimlet works. The SIP Zimlet "says" SIP, but it's not, it's using a reworked Asterisk piece which calls for fields SIP does not use, and is therefore, useless with a standards-based SIP system. (IMHO it should be renamed to reflect Asterisk and not the SIP protocol).

Anyone have any feedback or success with SIP using the Zimlet?

[netjunkie]

Hi,

Anyone know where this zimlet can be downloaded. Its not on the list in the gallery.

[phoenix]

If you're talking about the Asterisk zimlet then it's located in /opt/zimbra/zimlets-extra on your server.

[techieg]

Open standards is the way to go. When it comes to telephony, I will suggest it is better and more interoperable to implement an inbuilt click-to-call based on SIP rather than something suited to a single system such as what has been done with the "Asterisk Zimlet". You need to realize that different users/businesses run several different IPBXs that are not Asterisk-based and some even avoid Asterisk due to lack of scalability amongst other things. So it is a no-brainer to;

1. Rework the click-to-call feature as an inbuilt feature of Zimbra rather than a Zimlet
2. Rework the zimlet to initiate click-to-call based on the SIP format. That is, to send the called party's number as well as the calling party's number and the bridge to whatever IPBX the user may be using in a SIP format to initiate and bridge the call between both numbers.

With such an implementation, SIP will receive these credentials and do just that, and everyone's choice of PBX will be addressed rather than having a clunky "Asterisk-style" module. Asterisk users are stuck in the "Asterisk module" way of thinking rather than thinking open standard. This was the same problem with the Asterisk Skype module until Skype realized that if they wanted to reach farther they had no choice but to go open standard, so they now have SIP interop with Skype for SIP Beta - Skype .

We are a 501(c)(3) technology non-profit (Open Software Foundation - The Foundation for Information & Communications Technology.™) currently evaluating telephony/UC engines to derive a robust open source UC solution in-line with Zimbra (similar to and beyond how OCS augments Exchange) with a rich Ajax UI (perhaps YUI). We are not at all entertaining the idea of Asterisk because we need the system to be scalable and interoperate well enterprise-wise not just SMB, amongst other things (we certainly need Zimbra developers as well as others to join/work with us on this). Freeswitch seems more like what may fit this model because other engines seem to be a derivative of the very limited Asterisk.

Unless you only plan to reach a small group of users or small office rather than enterprise users open standard should be the default model of development and not some patch work to only fit a certain system.

[grazman]

[QUOTE=techieg;134250]Open standards is the way to go. When it comes to telephony, I will suggest it is better and more interoperable to implement an inbuilt click-to-call based on SIP rather than something suited to a single system such as what has been done with the "Asterisk Zimlet". You need to realize that different users/businesses run several different IPBXs that are not Asterisk-based and some even avoid Asterisk due to lack of scalability amongst other things.

<snip>

Since the guys at Digium sponsored or wrote the Zimlet, it's not like it was provided by Zimbra. At least they did that.

At the same time the marketing or naming of it should have been "asterisk" and not "sip". Call it what it is, I feel like it's a deceptive practice that I've seen with guys who get to big for their britches. They have never been my choice for PBX stuff, too clunky and maintenance heavy.

While seeing a sip based zimlet has a lot of appeal to me, I don't see a whole lot of people asking to help write one. When there is one maybe I'll do more with Zimbra once I can understand their licensing again.

[techieg]

[QUOTE=grazman;134255]

Originally Posted by techieg

Open standards is the way to go. When it comes to telephony, I will suggest it is better and more interoperable to implement an inbuilt click-to-call based on SIP rather than something suited to a single system such as what has been done with the "Asterisk Zimlet". You need to realize that different users/businesses run several different IPBXs that are not Asterisk-based and some even avoid Asterisk due to lack of scalability amongst other things.

<snip>

Since the guys at Digium sponsored or wrote the Zimlet, it's not like it was provided by Zimbra. At least they did that.

At the same time the marketing or naming of it should have been "asterisk" and not "sip". Call it what it is, I feel like it's a deceptive practice that I've seen with guys who get to big for their britches. They have never been my choice for PBX stuff, too clunky and maintenance heavy.

While seeing a sip based zimlet has a lot of appeal to me, I don't see a whole lot of people asking to help write one. When there is one maybe I'll do more with Zimbra once I can understand their licensing again.

Of course you have good points, which I understand. Only if Digium is not stuck on that mindset that lousy modules will do the job they could have come up with a SIP solution instead, which is why they cannot penetrate telephony markets beyond small groups of users where they currently are.

Since not everyone is able to write code or program/develop applications, I should probably add an RFE for SIP-based click-to-call so we can vote on it for Zimbra dev team to know how much this is needed. So we can all finally ditch this "asterisk-zimlet" and go open standard SIP instead thus making Zimbra more enterprise interoperable.

[techieg]

The RFE has been created here; Bug 37065 &ndash; SIP_based Click-to-Call

By the way, does anyone know what IMAP and HTTP/S servers Yahoo mail and Gmail use? We assume they must be very robust due to millions of concurrent user access and would like to use those for our UC solution.

[techieg]

I have browsed these areas and I don't think such an information is there. They mainly mention general settings without naming what they use to host the services.